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