Detlef

Ha, everything is slower in Mexico... I'm still strugelling to solve my problem but for what it is worth: my local IT guy in Mexico contacted Telmex and the statement from their rep was that Telmex or Mexico gov does not permit VoIP in general. So it is somehow a grey zone while many internet subscriber still use VoIP it may cause the ISP to terminate the internet contract completely. I also found many reports on the net that south america and especially Mexico ISP not totally block VoIP ports but do regulate bandwidth and disturb connection reliability. Well, that would explain my one way audio drop out... As of the Telmex statement the ones that do provide VoIP do it mainly via VPNs to tunnel those data out of Mexico. In that case I can stop looking for a local ITSP in Mexico for right now and start tunneling VoIP data to my pbxnsip systems in the US...

I was just wondering if anyone knows a decent ITSP in Mexico that I could use with PBXnSIP?? Currently I use Callcentric for all our offices but I keep getting complains from our Mexico office that with calls using Callcentric one channel drops out and the caller is left with one way audio. I assume it is a distance/Callcentric problem with going from Mexico to Callcentric because I also have VPN connections over the same internet conenction inbetween our own offices with PBXnSIP and those rarely loose one side of the audio transmission. In order to evaluate that I was thinking to setup a provider that is located in Mexico to see if that would solve my problem that I am having with Callcentric. Regards, Detlef

Well, I am running always the latest releases in three locations in the US and Mexico and so far I have not expirienced any crashes or major bugs that would prevent me from using the latest version. But I am the mean administrator and don't have to please customers (hehehe). Overall I am very happy with pbxnsip - no major issues and always quick responses and fixes to little problems I discovered so far.

That with the IVR node seems to work: So now my auto attendants transfers interoffice calls originating from PSTN or IP to silent IVR nodes instead of the direct auto attendant in the other office. [8] 2008/02/15 14:27:17: Found 200 in address book of domain localhost [5] 2008/02/15 14:27:17: Trunk VoIP VA sends call to 181 [8] 2008/02/15 14:27:17: Play recordings/ivr21.wav [6] 2008/02/15 14:27:17: Received DTMF 1 [7] 2008/02/15 14:27:17: Attendant: Set language to first language en [8] 2008/02/15 14:27:17: Play recordings/att2.wav space20 But with just "!E!100!" in the DTMF match field it gets stuck there with dead silence after it receives the not wanted DTMF 1. I put in the Timeout field a 1sec - that didnt change anything. The only thing that helped is a "!E!100! ![0-9]!100!" in the DTMF match field and it connects the call directly to the auto attendant for any digit received - even skipping the 1 sec silent wav if its a callcentric call.

So you mean Callcentric is faster/longer detecting and transmitting the DTMF than the Grandstream gateway? Hmm, I was trying to simlulate that by pressing and holding the key but all the phones I have here right now only send a short DTMF tone even if I hold the key. How does the PBX evaluate those signals? I have inband DTMF detection turned off in both systems, so it must get it via a SIP message? I dont know how that works but isnt that just one message saying the caller pressed 1 and not continously sending that tone? In the log you can see, it all happens in the same second at the receiving pbxnsip, transfer and detecting DTMF and then skipping the default auto attendant recording and then waiting for the timeout. So with the IVR node I would first send the call from the auto attendant to a local IVR node and that one would transfer the call to the other system after playing a blank 1 sec wav ?

I am running 2 pbxnsip 2.1.6.2446 (one in VA and one in GA). Both are interconnected with SIP Gateway trunks. GA uses extension 100-199 and VA uses extensions 200-299. All incoming calls go to the auto attendants 100 or 200. The auto attendants have a user input set to forward calls to the other system, so if a caller presses 1 in VA the auto attendant forwards the call to ext. 100 - which is located in GA and is sent there via the gateway trunk and dial plan. If someone calls in GA and presses the 2 the auto attendant there connects the caller to extension 200 - which is the VA auto attendant. That works so far very good if the call comes in over our Grandstream PSTN gateway. Now I added inward numbers from callcentric and the forward to the other's system auto attendant does not work. In fact it forwards the call to the other system and also sends for example the digit 1 that is used in VA to connect the call to the auto attendant in GA additional to GA. Of course the auto attendant in GA expects a 3 digit 1xx extension number and comes back with the message that the extension doesnt exist. How do I prevent the VoIP call from callcentric to send this additional digit that is used to initiate the transfer to the other system also to the other system? Below I added the two logs - first the VoIP call and second the working PSTN call: VA - Call from CallCentric to PBX [5] 2008/02/15 09:41:08: Trunk VoIP CC sends call to 200 [7] 2008/02/15 09:41:08: Attendant: Set language to first language en [8] 2008/02/15 09:41:08: Play recordings/att8.wav space20 [6] 2008/02/15 09:41:12: Received DTMF 1 [5] 2008/02/15 09:41:12: Dialplan default: Match 100@localhost to <sip:100@192.168.0.220;user=phone> on trunk VoIP GA [5] 2008/02/15 09:41:12: Using <sip:17574685765@66.193.176.35>;tag=3412075315-860014 as redirect from GA - Receiving PBX [7] 2008/02/15 09:41:57: UDP: Opening socket on port 63732 [7] 2008/02/15 09:41:57: UDP: Opening socket on port 63733 [5] 2008/02/15 09:41:57: Identify trunk (IP address/port and domain match) 1 [6] 2008/02/15 09:41:57: Sending RTP for c3fa37dc@pbx#02004e4d5f to 192.168.104.220:54466 [8] 2008/02/15 09:41:57: Found 200 in address book of domain localhost [5] 2008/02/15 09:41:57: Trunk VoIP VA sends call to 100 [7] 2008/02/15 09:41:57: Attendant: Set language to first language en [8] 2008/02/15 09:41:57: Play recordings/att2.wav space20 [6] 2008/02/15 09:41:57: Received DTMF 1 [8] 2008/02/15 09:42:00: Attendant: Timeout (press) [8] 2008/02/15 09:42:00: Play audio_en/aa_not_existing.wav space20 ===================== VA - Call from PSTN to PBX [5] 2008/02/15 09:45:59: Trunk PSTN VA sends call to 200 [7] 2008/02/15 09:45:59: Attendant: Set language to first language en [8] 2008/02/15 09:45:59: Play recordings/att8.wav space20 [7] 2008/02/15 09:45:59: Set packet length to 20 [6] 2008/02/15 09:46:05: Received DTMF 1 [5] 2008/02/15 09:46:05: Dialplan default: Match 100@localhost to <sip:100@192.168.0.220;user=phone> on trunk VoIP GA [5] 2008/02/15 09:46:05: Using "IMS GEAR, INC " <sip:7574685765@localhost> as redirect from [8] 2008/02/15 09:46:05: Play audio_moh/noise.wav GA - Receiving PBX [7] 2008/02/15 09:46:49: UDP: Opening socket on port 52728 [7] 2008/02/15 09:46:49: UDP: Opening socket on port 52729 [5] 2008/02/15 09:46:49: Identify trunk (IP address/port and domain match) 1 [6] 2008/02/15 09:46:49: Sending RTP for 73767b3c@pbx#452ad5f3c4 to 192.168.104.220:50578 [8] 2008/02/15 09:46:49: Found 200 in address book of domain localhost [5] 2008/02/15 09:46:49: Trunk VoIP VA sends call to 100 [7] 2008/02/15 09:46:49: Attendant: Set language to first language en [8] 2008/02/15 09:46:49: Play recordings/att2.wav space20

Hi Paul, I have to add that even with the domains set to unlimited you still get the other failure during startup but as u said the license is valid again: [4] 2008/02/13 17:22:03: Translation item dom_ext2.htm#default#sp not found

Do you already know if you will have a PBX version 2.1.0.2116 that will support this historic on hold procedures again?? If I wouldn't have bought 40 of those stupid Aastra phones over the past couple months I wouldnt be so desperate

I would really appreciate that, currently I am stuck with the new PBX not supporting it any more and Aastra not knowing if or when they update their phones... the pain ist that it is silent to the caller if you put someone on hold with the Aastra phone and they think the line dropped and mostly hang up. I guess their latest firmware is used in all 4 of those phones listed in the release info? FC-000032-01-11 480i FC-000040-00-11 480iCT FC-000046-01-11 9133i FC-000058-01-11 9112i

Nice, I got a reply from Aastra. They have this problem on their feature request list... hopefully not since 5 years!! EDIT: Oh, just got an accurate estimate - its even worse than 5 years, they have no clue when they will update their phones:

hehehe, but I did anyway Well, I think it all depends on your requirements. In my case we are not a business that constanly is talking to customers over the phone. I would guess that over 90% of our calls are business internal but to world wide locations. In that case we accept lower quality (which didn't really happen so far) and failing calls (real minimal) for business internal calls over the fact that our calls are basically free. Our main goal was reducing cost. Our PSTN gateway acts in this scenario as safe way if other things fail and for all incoming calls. Running everything outgoing over a cheap cable modem here in the US is working much better as expected.

For you SIP Trunk you dont need a gateway. Those Gateway devices are meant to adapt all the differnet available PSTN (public switched telephone network) connections into an IP enviroment. It doesnt matter if its analog or digital - its just not an IP network type. I think in the presentation the recomendation to have a gateway to a PST Network (regular phone line) is more for safety and as backup if the internet fails.

oops, the GWX410x is pure analog PSTN... havent used any T1,E1,BRI VoIP products so I cant give you any help there. I am currently looking for an ISDN gateway into a phone system in Germany but that is not in my hands so I am not involved too much in the selection process.

I just put the latest and greatest firmware on the phone that was available from their website... AASTRA TELECOM INC. June 2007 Generic SIP Firmware 1.4.2.1081 GA Release. FC-000032-01-11 480i FC-000040-00-11 480iCT FC-000046-01-11 9133i FC-000058-01-11 9112i Maybe I can have them change it if I contact their customer support and ask for the correct ON HOLD procedure? What would I need to tell them to see if they would change it to the up-to-date way? EDIT: I just sent them an email... lets see what they will say

I used an AudioCodes Gateway in the past and thought that thing had way too many options and made it complicated to find the right settings spread out over too many pages in the web admin interface. Also I didnt have the greatest expirence with the AudioCodes support - they never responded. The Grandstream GXW has much less settings, their customer support at least responds and downloads are available from their web site without signing up and waiting for an answer that never comes. Still I think the Grandstream might not be the best model but it does what I want and its not too hard to configure. So far it works with the PBXnSIP together without any problems but I dont have any complicated feature demands for that thing either. All I want is to receive and make calls and get caller ID on PSTN lines into the PBX and that works without any complains so far. Detlef

This is the logfile I see as soon as I press the HOLD button on the Aastra phone during a connected call with the X-Lite softphone: 192.168.104.222 = Aastra Phone 192.168.104.129 = X-Lite Softphone 192.168.104.220 = PBXnSIP [9] 2007/10/10 08:55:20: SIP Rx udp:192.168.104.222:5060: INVITE sip:101@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bKdd007bf64 Max-Forwards: 70 Content-Length: 266 To: "X-Lite" <sip:150@localhost>;tag=43086 From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a Call-ID: e4afabe4@pbx CSeq: 69380235 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Aastra <sip:101@192.168.104.222> Supported: replaces User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1152796867 IN IP4 192.168.104.222 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 3000 RTP/AVP 0 8 18 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [9] 2007/10/10 08:55:20: Resolve destination 2684: a udp 192.168.104.222 5060 [9] 2007/10/10 08:55:20: Resolve destination 2684: udp 192.168.104.222 5060 [9] 2007/10/10 08:55:20: SIP Tx udp:192.168.104.222:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bKdd007bf64 From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a To: "X-Lite" <sip:150@localhost>;tag=43086 Call-ID: e4afabe4@pbx CSeq: 69380235 INVITE Contact: <sip:101@192.168.104.220:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.0.2115 Content-Type: application/sdp Content-Length: 275 v=0 o=- 34981 34981 IN IP4 192.168.104.220 s=- c=IN IP4 192.168.104.220 t=0 0 m=audio 64154 RTP/AVP 0 8 18 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2007/10/10 08:55:20: SIP Rx udp:192.168.104.222:5060: ACK sip:101@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK31b41b11a Max-Forwards: 70 Content-Length: 0 To: "X-Lite" <sip:150@localhost>;tag=43086 From: "Aastra 9133i" <sip:101@localhost>;tag=7fb09db3322e24a Call-ID: e4afabe4@pbx CSeq: 69380235 ACK Contact: Aastra <sip:101@192.168.104.222> User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

Well, for me I think its a safe backup in the case that our VoIP fails. Also I use the PSTN lines as dial-in numbers over a Grandstream GWX4108 gateway and for free local and toll-free calls, where our callcentric.com VoIP provider would charges us for. Also 911 is a consideration if the VoIP provider does not provide this function. Additionaly I need the PSTN for burglar and fire alarm anyway, so why not use it for some calls as well. Also our fax runs currently over a regular line - that way I didnt have to mess with the T.38 in the beginning of our VoIP rollout. We have about 20 extension each in two US and one Mexico location. Since we have Mexico involved and don't count on a stable internet connection - if this fails we use PSTN with a simple dialplan entry. For example if a user dials 99xxxx then the PBX uses PSTN. Due to the VoIP we cut the number of PSTN lines in half because most calls use now VoIP. So far we are happy with this solution. It safes us alot of money for calls but allows us to go back any time to call out on PSTN if we encounter problems, its not the cheapest setup looking at monthly cost but I think with the best functionality.

Ok done, upgraded to .2115 and the 9133i Aastra phones still do not initiate MOH when putting someone on hold or parking a call. Is someone using 9133i phones who has it working? Maybe we could exchange config files??? Would be really appreciated!! Detlef

Will do tonight, but since I upgraded to any previous 2.1.0.xxxx version the 9133i phones here quit doing MOH. Going back to 2.0.3.1715 and the MOH always started working again. What I dont understand is that the X-Lite softphone does it correct. I went to the newst Aastra firmware available for download but no change - still no MOH. Below my MAC.CFG everything else on the Aastra settings is left to default: download protocol: TFTP tftp server: pbx.ims-va.com auto resync mode: 3 auto resync time: 03:00 time server disabled: 0 time server1: dc.ims-va.com sip digit timeout: 3 sip dial plan: "x+#|xx+*" sip mode: 0 sip proxy ip: pbx.ims-va.com sip proxy port: 5060 sip registrar ip: pbx.ims-va.com sip registrar port: 0 sip registration period: 3600 sip screen name: User Name sip user name: 101 sip display name: User Name sip auth name: 101 sip password: xxxxxx sip vmail: *97 directory 1: companylist.csv directory 2: externallist.csv

Since I upgraded to PBX 2.1.0.xxxx my MOH is not working no more. Now after some testing I think it is the Aastra phone causing this. If I use an extension with a softphone X-Lite and call another extension with an Aastra 9133i phone and put the call on hold from the X-Lite softphone then the Aastra plays music on hold (from default file). Do I do it the oposite way and put the call on hold from the Aastra 9133i phone then its silent and no MOH plays. The same happens when I call in over the PSTN gateway. If I call the extension with the X-Lite softphone and put that call on hold from the softphone then the caller hears music on hold. If I call the extension with the Aastra phone and put it on hold then its silent to the caller over the PSTN line. Is that a setting in the Aastra phones that I need to correct? It was working with PBXnSIP 2.0.3.1715 and quit with the new 2.1.0.xxxx versions. Since the X-Lite softphone does it correctly I think it has to do with the Aastra phones themself. Currently I am running PBXnSIP 2.1.0.2114 and the Aastra 9133i is firmware 1.4.2.1081 Detlef

Yes it actually does work now. The only other thing I changed yesterday was the codec preference under the PBXnSIP settings in the ports screen. There I had one system configured to the default 0 8 18 2 3 and the other one to 18 0 8 2 3. I have both systems licenses with 10 G.729 codecs. So both should have been able to handle the preference 18 from the one system. Anyway, I changed both back to the default 0 8 18 2 3. Dont ask me if that made the difference. Detlef

Now beats me, I changed some settings around in the GS gateway. Some wait for dialtone timing which should affect my connection problems between the two PBXnSIP but I also added in the channels page for each channel a number in the field as suggested here in some other post about GXW410x... and wonder I all a sudden get audio to my PBXnSIP in Mexico. I dont see any difference in the logfile either... hmmm Working incoming PSTN Call forwarded to second PBXnSIP: ======================================== [7] 2007/10/08 17:57:27: UDP: Opening socket on port 63454 [7] 2007/10/08 17:57:27: UDP: Opening socket on port 63455 [5] 2007/10/08 17:57:27: Identify trunk (IP address and domain match) 6 [7] 2007/10/08 17:57:27: Set packet length to 20 [6] 2007/10/08 17:57:27: Sending RTP for 58806270b5d787c5@192.168.104.26#3cf4ad98e7 to 192.168.104.26:5020 [5] 2007/10/08 17:57:27: Trunk PSTN Cox sends call to 100 [7] 2007/10/08 17:57:27: Received call from cell phone 757xxxxxxx [8] 2007/10/08 17:57:27: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50 [7] 2007/10/08 17:57:27: Set packet length to 20 [6] 2007/10/08 17:57:32: Received DTMF 1 [8] 2007/10/08 17:57:32: Play audio_en/ex_enter_access_code.wav [8] 2007/10/08 17:57:34: Play space20 [6] 2007/10/08 17:57:34: Received DTMF 9 [6] 2007/10/08 17:57:35: Received DTMF 9 [6] 2007/10/08 17:57:36: Received DTMF 9 [8] 2007/10/08 17:57:36: Play audio_en/ex_enter_number.wav [6] 2007/10/08 17:57:38: Received DTMF 3 [6] 2007/10/08 17:57:39: Received DTMF 8 [6] 2007/10/08 17:57:40: Received DTMF 0 [6] 2007/10/08 17:57:40: Received DTMF # [5] 2007/10/08 17:57:41: Dialplan: Match 380@localhost to <sip:380@192.168.108.220;user=phone> on trunk VoIP MX [5] 2007/10/08 17:57:41: Using "User Name" <sip:757xxxxxxx@localhost> as redirect from [8] 2007/10/08 17:57:41: Play audio_moh/noise.wav [7] 2007/10/08 17:57:41: UDP: Opening socket on port 55478 [7] 2007/10/08 17:57:41: UDP: Opening socket on port 55479 [7] 2007/10/08 17:57:41: Call 93a2ec14@pbx#7094: Clear last INVITE [6] 2007/10/08 17:57:41: Sending RTP for 93a2ec14@pbx#7094 to 192.168.108.220:59096 [7] 2007/10/08 17:57:41: 93a2ec14@pbx#7094: RTP pass-through mode [7] 2007/10/08 17:57:41: 58806270b5d787c5@192.168.104.26#3cf4ad98e7: RTP pass-through mode [7] 2007/10/08 17:57:51: Other Ports: 2 [7] 2007/10/08 17:57:51: Call Port: 93a2ec14@pbx#7094 [7] 2007/10/08 17:57:51: Call Port: 97829f62e3d24894@192.168.104.26#9fb3705bc9 [7] 2007/10/08 17:57:51: Call 93a2ec14@pbx#7094: Clear last request [5] 2007/10/08 17:57:51: BYE Response: Terminate 93a2ec14@pbx

This is what the GS GXW4108 logs in DEBUG mode if I call in: 10-08-2007 17:14:19 Sess: 1 INVITE sip:100@192.168.104.220 SIP/2.0 Via: SIP/2.0/UDP 192.168.104.26:5064;branch=z9hG4bK140724266952cdb2 From: "Cell Phone VA"<sip:757xxxxxxx@192.168.104.220>;tag=5120b893f992da17 To: <sip:100@192.168.104.220> Contact: <sip:192.168.104.26:5064> Supported: replaces, timer, path Call-ID: cf40f4173c6140e0@192.168.104.26 CSeq: 31868 INVITE User-Agent: Grandstream GXW4108 (HW 0001, Ch:1) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 290 v=0 o=system 8001 8000 IN IP4 192.168.104.26 s=SIP Call c=IN IP4 192.168.104.26 t=0 0 m=audio 5008 RTP/AVP 0 8 18 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 10-08-2007 17:14:19 Sess: 1 ACK sip:100@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.26:5064;branch=z9hG4bKeb37fb8018c5b932 From: "Cell Phone VA"<sip:757xxxxxxx@192.168.104.220>;tag=5120b893f992da17 To: <sip:100@192.168.104.220>;tag=39c9da9605 Contact: <sip:192.168.104.26:5064;user=phone> Call-ID: cf40f4173c6140e0@192.168.104.26 CSeq: 31868 ACK User-Agent: Grandstream GXW4108 (HW 0001, Ch:1) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 10-08-2007 17:14:28 Sess: 1 BYE sip:100@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.26:5064;branch=z9hG4bK8b2096112a327c03 From: "Cell Phone VA"<sip:757xxxxxxx@192.168.104.220>;tag=5120b893f992da17 To: <sip:100@192.168.104.220>;tag=39c9da9605 Call-ID: cf40f4173c6140e0@192.168.104.26 CSeq: 31869 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:1) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0

I did some more try and error. Trying to dial in with my regular cell phone over the PSTN and the Grandstream gateway the audio fails to transmitted to the second pbx system. If I try to call in over a callcentric account the audio works. Comparing the logs the only difference I see is this: Fails - Calling in over Grandstream PSTN Gateway: [5] 2007/10/08 14:16:22: Using "User Name (Cell)" <sip:757xxxxxxx@localhost> as redirect from Works - Calling in over Callcentric: [5] 2007/10/08 13:52:42: Using "User Name" <sip:1777xxxxxxx@callcentric.com>;tag=781da70e as redirect from It puts on the grandstream call a @localhost as redirect from rather than the real hostname or IP. Could that be the reason why I dont hear any audio? As attachment both complete logs... Fails - Call over PSTN Gateway: =============================== [7] 2007/10/08 14:16:09: UDP: Opening socket on port 62650 [7] 2007/10/08 14:16:09: UDP: Opening socket on port 62651 [5] 2007/10/08 14:16:09: Identify trunk (IP address and domain match) 6 [7] 2007/10/08 14:16:09: Set packet length to 20 [6] 2007/10/08 14:16:09: Sending RTP for 74624ba1f9843062@192.168.104.26#59b97d943e to 192.168.104.26:5008 [5] 2007/10/08 14:16:09: Trunk PSTN Cox sends call to 100 [7] 2007/10/08 14:16:09: Received call from cell phone 757xxxxxxx [8] 2007/10/08 14:16:09: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50 [7] 2007/10/08 14:16:09: Set packet length to 20 [5] 2007/10/08 14:16:10: Registration on trunk 5 (VoIP GER) failed. Retry in 60 seconds [6] 2007/10/08 14:16:12: Received DTMF 1 [8] 2007/10/08 14:16:12: Play audio_en/ex_enter_access_code.wav [8] 2007/10/08 14:16:14: Play space20 [6] 2007/10/08 14:16:15: Received DTMF 9 [6] 2007/10/08 14:16:16: Received DTMF 9 [6] 2007/10/08 14:16:17: Received DTMF 9 [8] 2007/10/08 14:16:17: Play audio_en/ex_enter_number.wav [6] 2007/10/08 14:16:20: Received DTMF 3 [6] 2007/10/08 14:16:21: Received DTMF 8 [6] 2007/10/08 14:16:22: Received DTMF 0 [6] 2007/10/08 14:16:22: Received DTMF # [5] 2007/10/08 14:16:22: Dialplan: Match 380@localhost to <sip:380@192.168.108.220;user=phone> on trunk VoIP MX [5] 2007/10/08 14:16:22: Using "User Name (Cell)" <sip:757xxxxxxx@localhost> as redirect from [8] 2007/10/08 14:16:22: Play audio_moh/noise.wav [7] 2007/10/08 14:16:22: UDP: Opening socket on port 51824 [7] 2007/10/08 14:16:22: UDP: Opening socket on port 51825 [7] 2007/10/08 14:16:23: Call 63ce148a@pbx#38760: Clear last INVITE [6] 2007/10/08 14:16:23: Sending RTP for 63ce148a@pbx#38760 to 192.168.108.220:56130 [7] 2007/10/08 14:16:23: 63ce148a@pbx#38760: RTP pass-through mode [7] 2007/10/08 14:16:23: 74624ba1f9843062@192.168.104.26#59b97d943e: RTP pass-through mode [7] 2007/10/08 14:16:33: Other Ports: 3 [7] 2007/10/08 14:16:33: Call Port: 09a07909e67a241414771b275404cb9f@192.168.104.235#186da2d5ab [7] 2007/10/08 14:16:33: Call Port: 63ce148a@pbx#38760 [7] 2007/10/08 14:16:33: Call Port: 9351a6ed@pbx#34231 [7] 2007/10/08 14:16:33: Call 63ce148a@pbx#38760: Clear last request [5] 2007/10/08 14:16:33: BYE Response: Terminate 63ce148a@pbx [7] 2007/10/08 14:16:33: Other Ports: 2 [7] 2007/10/08 14:16:33: Call Port: 09a07909e67a241414771b275404cb9f@192.168.104.235#186da2d5ab [7] 2007/10/08 14:16:33: Call Port: 9351a6ed@pbx#34231 Works - Call over Callcentric: ============================== [7] 2007/10/08 13:52:32: UDP: Opening socket on port 61862 [7] 2007/10/08 13:52:32: UDP: Opening socket on port 61863 [5] 2007/10/08 13:52:32: Identify trunk (line match) 4 [6] 2007/10/08 13:52:32: Sending RTP for NjUxZTFhZWQxZjFkYWE4ZDE1OWUyNDgxOGU3M2Y0ZGQ.#c8a9483aee to 204.11.192.22:63674 [5] 2007/10/08 13:52:32: Trunk VoIP CC sends call to 100 [7] 2007/10/08 13:52:32: Received call from cell phone 1777xxxxxxx [8] 2007/10/08 13:52:32: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50 [6] 2007/10/08 13:52:34: Received DTMF 1 [8] 2007/10/08 13:52:34: Play audio_en/ex_enter_access_code.wav [6] 2007/10/08 13:52:36: Received DTMF 9 [6] 2007/10/08 13:52:36: Received DTMF 9 [6] 2007/10/08 13:52:37: Received DTMF 9 [8] 2007/10/08 13:52:37: Play audio_en/ex_enter_number.wav [6] 2007/10/08 13:52:41: Received DTMF 3 [6] 2007/10/08 13:52:41: Received DTMF 8 [6] 2007/10/08 13:52:42: Received DTMF 0 [6] 2007/10/08 13:52:42: Received DTMF # [5] 2007/10/08 13:52:42: Dialplan: Match 380@localhost to <sip:380@192.168.108.220;user=phone> on trunk VoIP MX [5] 2007/10/08 13:52:42: Using "User Name" <sip:1777xxxxxxx@callcentric.com>;tag=781da70e as redirect from [8] 2007/10/08 13:52:42: Play audio_moh/noise.wav [7] 2007/10/08 13:52:42: UDP: Opening socket on port 62364 [7] 2007/10/08 13:52:42: UDP: Opening socket on port 62365 [7] 2007/10/08 13:52:42: Call 4408d49c@pbx#5176: Clear last INVITE [6] 2007/10/08 13:52:42: Sending RTP for 4408d49c@pbx#5176 to 192.168.108.220:54286 [7] 2007/10/08 13:52:42: 4408d49c@pbx#5176: RTP pass-through mode [7] 2007/10/08 13:52:42: NjUxZTFhZWQxZjFkYWE4ZDE1OWUyNDgxOGU3M2Y0ZGQ.#c8a9483aee: RTP pass-through mode [7] 2007/10/08 13:52:47: Other Ports: 1 [7] 2007/10/08 13:52:47: Call Port: 4408d49c@pbx#5176 [7] 2007/10/08 13:52:47: Call 4408d49c@pbx#5176: Clear last request [5] 2007/10/08 13:52:47: BYE Response: Terminate 4408d49c@pbx

Uh, that sounds easy, I could have had that idea. I'll try it if I get a chance. As far as PBXnSIP version for the unicast paging I am using the 2.1.0.2114 currently. But I havent looked at the detailed log file why I dont get any audio with the unicast paging.