shopcomputer

Outbound Proxy on the trunk is set to east.ga.broadvox.net which is successfully resolving to 64.152.60.75 the from address on this call.

I am trying to figure out what I am doing wrong, I am using the tel:alias for the first time for a DID block provided by the carrier. I set alias names to tel:3474244022 for extension 22, it is not working, and this is what I see in the logs. If I set the trunk to send all calls to the auto-attendands extension then this new DID block does work. [0] 2008/02/08 09:37:28: SIP Rx udp:64.152.60.75:5060: INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222> Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:9174170964@64.152.60.75:5060> P-Asserted-Identity: <sip:9174170964@64.152.60.75:5060> Supported: timer Session-Expires: 1800 Min-SE: 90 Content-Length: 279 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 3495 31360 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.164 t=0 0 m=audio 8280 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2008/02/08 09:37:28: UDP: Opening socket on port 50448 [7] 2008/02/08 09:37:28: UDP: Opening socket on port 50449 [0] 2008/02/08 09:37:28: SIP Tx udp:64.152.60.75:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 INVITE Content-Length: 0 [7] 2008/02/08 09:37:28: Set packet length to 20 [6] 2008/02/08 09:37:28: Sending RTP for 1712050395_129292298@64.152.60.75#adad1c9d5d to 64.152.60.164:8280 [5] 2008/02/08 09:37:28: Received incoming call without trunk information and user has not been found [7] 2008/02/08 09:37:28: Set packet length to 20 [0] 2008/02/08 09:37:28: SIP Tx udp:64.152.60.75:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 INVITE Contact: <sip:3474244022@64.113.246.222:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [0] 2008/02/08 09:37:28: Last message repeated 2 times [0] 2008/02/08 09:37:28: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 ACK Max-Forwards: 70 Content-Length: 0 [7] 2008/02/08 09:37:28: Other Ports: 2 [7] 2008/02/08 09:37:28: Call Port: 3c47348884cd-5rqcm28316wh#24f2d0d27e [7] 2008/02/08 09:37:28: Call Port: 3ed31811@pbx#28804 [0] 2008/02/08 09:37:28: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d807057fc936360 From: <sip:9174170964@64.152.60.75>;tag=gK0b676a49 To: <sip:3474244022@64.113.246.222>;tag=adad1c9d5d Call-ID: 1712050395_129292298@64.152.60.75 CSeq: 11947 ACK Max-Forwards: 70 Content-Length: 0 [0] 2008/02/08 09:37:29: SIP Rx udp:64.152.60.75:5060: INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222> Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:9174170964@64.152.60.75:5060> P-Asserted-Identity: <sip:9174170964@64.152.60.75:5060> Supported: timer Session-Expires: 1800 Min-SE: 90 Content-Length: 279 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 15387 5494 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.71 t=0 0 m=audio 12958 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2008/02/08 09:37:29: UDP: Opening socket on port 53930 [7] 2008/02/08 09:37:29: UDP: Opening socket on port 53931 [0] 2008/02/08 09:37:29: SIP Tx udp:64.152.60.75:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 INVITE Content-Length: 0 [7] 2008/02/08 09:37:29: Set packet length to 20 [6] 2008/02/08 09:37:29: Sending RTP for 1712050397_64827812@64.152.60.75#cb5aae4b1a to 64.152.60.71:12958 [5] 2008/02/08 09:37:29: Received incoming call without trunk information and user has not been found [7] 2008/02/08 09:37:29: Set packet length to 20 [0] 2008/02/08 09:37:29: SIP Tx udp:64.152.60.75:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 INVITE Contact: <sip:3474244022@64.113.246.222:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [0] 2008/02/08 09:37:29: Last message repeated 2 times [0] 2008/02/08 09:37:29: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 ACK Max-Forwards: 70 Content-Length: 0 [7] 2008/02/08 09:37:29: Other Ports: 2 [7] 2008/02/08 09:37:29: Call Port: 3c47348884cd-5rqcm28316wh#24f2d0d27e [7] 2008/02/08 09:37:29: Call Port: 3ed31811@pbx#28804 [0] 2008/02/08 09:37:29: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8275c7a86162df From: <sip:9174170964@64.152.60.75>;tag=gK0b676b52 To: <sip:3474244022@64.113.246.222>;tag=cb5aae4b1a Call-ID: 1712050397_64827812@64.152.60.75 CSeq: 24595 ACK Max-Forwards: 70 Content-Length: 0 [0] 2008/02/08 09:37:30: SIP Rx udp:64.152.60.75:5060: INVITE sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222> Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:9174170964@64.152.60.75:5060> P-Asserted-Identity: <sip:9174170964@64.152.60.75:5060> Supported: timer Session-Expires: 1800 Min-SE: 90 Content-Length: 279 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 9420 9690 IN IP4 64.152.60.75 s=SIP Media Capabilities c=IN IP4 64.152.60.164 t=0 0 m=audio 23172 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2008/02/08 09:37:30: UDP: Opening socket on port 55196 [7] 2008/02/08 09:37:30: UDP: Opening socket on port 55197 [0] 2008/02/08 09:37:30: SIP Tx udp:64.152.60.75:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 INVITE Content-Length: 0 [7] 2008/02/08 09:37:30: Set packet length to 20 [6] 2008/02/08 09:37:30: Sending RTP for 1712050399_42141136@64.152.60.75#050f7bfefc to 64.152.60.164:23172 [5] 2008/02/08 09:37:30: Received incoming call without trunk information and user has not been found [7] 2008/02/08 09:37:30: Set packet length to 20 [0] 2008/02/08 09:37:30: SIP Tx udp:64.152.60.75:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 INVITE Contact: <sip:3474244022@64.113.246.222:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [0] 2008/02/08 09:37:30: Last message repeated 2 times [0] 2008/02/08 09:37:30: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 ACK Max-Forwards: 70 Content-Length: 0 [7] 2008/02/08 09:37:30: Other Ports: 2 [7] 2008/02/08 09:37:30: Call Port: 3c47348884cd-5rqcm28316wh#24f2d0d27e [7] 2008/02/08 09:37:30: Call Port: 3ed31811@pbx#28804 [0] 2008/02/08 09:37:30: SIP Rx udp:64.152.60.75:5060: ACK sip:3474244022@64.113.246.222:5060 SIP/2.0 Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK0bB2d8453e5f2f4977f From: <sip:9174170964@64.152.60.75>;tag=gK0b676c59 To: <sip:3474244022@64.113.246.222>;tag=050f7bfefc Call-ID: 1712050399_42141136@64.152.60.75 CSeq: 12752 ACK Max-Forwards: 70 Content-Length: 0

I am running 2.1.5.2357, On Windows 2003 Web Edition, I will wait for the next version to be published to see if it is fixed.

I have a main auto attendant with a prompt 4# to go to a sub auto attendant with a directory listing. About 2 seconds after the sub listing starts playing the greeting, it gets interupted with a message please enter your entension number. If I dial the extension number of the sub auto attendant directly, without first going to the main auto attendant it works fine. All other menu's that do not have a # work fine. The time out on the main autoattendant is set to 60 seconds.

It seams to import OK to Level Platforms.

Never mind I figured it out, they show the display name field from the trunk.

Does anyone have any experience getting caller ID working with voicepulse connect. I put in the phone number in DID on the trunk, tried all caller ID presentation options, I get unknown number, or with several options user not found. In your extensions.conf file in the outgoing section before the dial statement you need to add in the SetCallerID line: ;[Extensions.conf config file] [outgoing] ;Enter the CID line here exten => _1NXXNXXXXXX,1,SetCallerID(5554443210) exten => _1NXXNXXXXXX,2,Dial(IAX2/UsErNaME:PaSSwoRD@gwiaxt01.voicepulse.com/${EXT EN}) exten => _1NXXNXXXXXX,103,Dial(IAX2/UsErNaME:PaSSwoRD@gwiaxt02.voicepulse.com/${E XTEN}) ;[/Extensions.conf config file] Most basic setups have the dial statement as position 1. In thise case you need to move dial to position 2, and change the failover dial to 103 (101+2). Keep in mind, the number you enter for CID needs to be a valid 10 digit number ([2-9]XX[2-9]XXXXXX). The CallerID name will NOT be sent, as this is currently a limitation of VOIP service

http://wiki.pbxnsip.com/index.php/Extension states there are 3 announcement modes, Anonymous Announcement, Name Announcement and Uploaded Announcement. I am not seeing the uploaded announcement option am I missing something?

They do not want to block the ability to dial the extension from the auto attendant, they just don't want the extension number published. They want The Ceo's freinds and VIP's to be able to dial his extension if they know it.

Is there any way to hide a user from the dial by name directory? My client does not want the Boss's extension number published in the directory. Also how do I handle a user who goes by the name Bob and Robert, he wants the dial by name directory to exept both.

We sell Broadvox, they were only selling to Enterprise until recently, now they started selling to the SMB through Vars only. Moishe Grunstein Tornado Computer Systems, Inc. www.nysolutions.com 212-400-7650

I am not having a sip traffic problem, I am having a problem getting this Tapi SP installed correctly.

I was trying to set this up in a office that has about 10 workstation, I tried 3 workstations with XP they did not work, I tried 1 Vista it did work, they do have some more workstations, however I did not bother trying it if the other 3 failed, they are all XP PRo SP2 and part of a domain if that makes a difference, I did try seting it up while logged in as a domain admin with the firewall turned off.

I was succesfull installing the Tapi SP on a Vista machine, however I tried it on several XP machines, although the setup says completed sucessfully, it appears to have not installed. The Tapi Service Provider was not added to the Tapi list. I tried manually adding it in the registry, which did add it to the list however it would not give me the configure button to configure the IP address of the PBX, etc.

Can I set a park button on the Aastra 53i or 57i, to automatically dial *85.. and *86... On the snom I use the buttons, park orbit to do this.

I there any way we can put a 1 or 2 second delay before a calls transfers to a direct destination. We have a direct destination for number 4 to a users extension, we have a user with ext. 45. When a user tries dialing 45, it transfers to the direct destination of 4, before the user gets to input the 5.

Snom finally took version 7 out of beta and released version 7.1.30 firmware today.

I have a user with 2 phones 1 Aastra 57i CT and 1 snom 360, when the user is talking on the Aastra and another user intercoms to him *90xx, the snom picks up the call the way it should, however the call talking on the Aastra gets terminated. Also any way to enter 2 mac addresses in the registration of a user for config of 2 phones? do I just seperate the macs with spaces?

The wilki says http://wiki.pbxnsip.com/index.php/Snom we need version 7.1.29, the latest I was able to find was 7.1.28, where can I download the 7.1.29, maybe it will improve my spearphone quality.

If someone leaves a voicemail and presses 0 while leaving the voicemail, the voicemail recording has the message the person left plus the transfer and their conversation with the operator, is there any way to turn that off? we only want the voicemail message to have the actual message they left, not them pressing 0 and talking to the operator. I got several complaints about this.

The story gets more strange, I woke up this morning, I try both the Broadvox and junction networks and they both work today.

I had broadvox working successfully for outbound for over a week, though it was not always working for inbound. I called in to support to try to fix inbound, and now outbound calls the destination number hears ringing and picks up and gets connected, however they don't hear anything. However we don't even hear a ringback. I was on the phone all day with both Broadvox and my PBXnsip distributor, they were both saying it was the others problem. though broadvox said they don't like our headers, it shows the calls coming from 34@localhost when calling from ext. 34, though they could not rule that that is the issue. We send around logs and captures and no resolution yet. I tried a juntion networks account which I use for a other PBX and I am getting the exact same thing. I tried callcentric as a test, and it seems to work, except for about 1 in 10 trys. We really made no changes on the PBX since yesterday when it worked succesfully and this morning when it was not working, I was talking to Broadvox support, however they say they made no changes they only put up a trace. The only change they told me to try try while I was on the phone with them was changing ringback from media to message 180, however we changed that back a minute later. The the mean time we changed the outgoing calls to use the audiocodes gateway we use for incoming. Any help would be greatly appreciated.

The speakerphone quality seems to be very bad on the Snom 360 using latest 7.1.28 firmware. We ruled out trunks being an issue, as we are having the problem even when intercoming to other extensions. It has alot of background noise, and sometimes an echo.

How call I disable the call waiting beeps on the snom 360 handsets Using 7.1.28 auto provisioned, users are complaining they can not hear the party they are talking to as the call waiting ring disrupt them.

The auto provisioning is working, however I don't see anything on that page regarding provisioning an intercom button.