Andy Hester
2003-Jun-29 18:59 UTC
[Asterisk-Users] Help! Problems talking to upstream switch
Hi, Please let me know if you have any ideas - I am taking wild guesses now.... Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP phones save a few fax machines. The customer has 7 digit unverified account codes on the trunks for billing purposes. The Problem: As I watch the console, I see calls coming in for exten "73" or "708" or "08" or "730" although most come in correctly (ie "7308"). My carrier has verified numerous times that they are sending 4 digits. I have 40 DID numbers that need to be routed and they are all in the 73xx range. I need to know anything that would cause my box randomly not to hear all 4 digits on occasion. Also, I have had trouble with people who dial out getting a congestion signal mainly on Long Distance numbers. The person would dial the number 4 or 5 times and get congestion then it might go through. Both of these conditions seem to be happening only about 10-20% of the time. What I have done: Moved a T100P card to its own IRQ to prevent problems with interrupts - Did not solve either issue. On the second try, got the carrier to change the way their switch chooses channels for incoming calls to prevent "glare" - This MAY have fixed the outgoing long distance issue as it seems to have gone away( although it doesn't seem logical to me that this would affect only LD) but did not fix incoming calls. Has anyone else had problems getting all of the digits that the Telco sends? Thanks, Andy Hester Consero
Steven Critchfield
2003-Jun-29 19:40 UTC
[Asterisk-Users] Help! Problems talking to upstream switch
Whats the sound quality like on the calls especially when multiple calls are going? On my home system, I had a problem that DTMF was sometimes not correctly recognized and would either dial incorrectly or not at all. It was evedent when dial tone was played that it crackled. Also when multiple calls where running each would start to sound like crap. This was later tracked down to a timing problem. Since you mention using a CAC with 2 ports on it and a router, I'm going to assume you have a ADIT 600. Make sure the Adit is set to take timing from the telco, and then make sure you are set to take your timing from it. Check to make sure each of the T1 ports a:1 and a:2 are set to take timing from a:1 if it is your telco port. This will keep you from slipping and causing potential problems. Come to think of it, there is a way to test this without bringing the T1s down. The Adit 600 has a show performance command, I may be wrong, but I'm sure it was performance, anyways it allows you to see slips and bipolar violations and a couple other stats. This was beneficial for me as the T100P didn't report problems but the Adit did. Home this helps. On Sun, 2003-06-29 at 20:59, Andy Hester wrote:> Hi, > Please let me know if you have any ideas - I am taking wild guesses now.... > Here is the situation: > > I put in Asterisk for a local customer. I have Fractional T-1 with 12 > Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a > chanel bank although it only has 2 outputs) in a CAC unit. The unit also > has a router card that runs the data side. My extensions are all SIP phones > save a few fax machines. The customer has 7 digit unverified account codes > on the trunks for billing purposes. > > The Problem: > > As I watch the console, I see calls coming in for exten "73" or "708" or > "08" or "730" although most come in correctly (ie "7308"). My carrier has > verified numerous times that they are sending 4 digits. I have 40 DID > numbers that need to be routed and they are all in the 73xx range. I need > to know anything that would cause my box randomly not to hear all 4 digits > on occasion. Also, I have had trouble with people who dial out getting a > congestion signal mainly on Long Distance numbers. The person would dial > the number 4 or 5 times and get congestion then it might go through. Both > of these conditions seem to be happening only about 10-20% of the time. > > What I have done: > > Moved a T100P card to its own IRQ to prevent problems with interrupts - Did > not solve either issue. > > On the second try, got the carrier to change the way their switch chooses > channels for incoming calls to prevent "glare" - This MAY have fixed the > outgoing long distance issue as it seems to have gone away( although it > doesn't seem logical to me that this would affect only LD) but did not fix > incoming calls. > > Has anyone else had problems getting all of the digits that the Telco sends? > > Thanks, > Andy Hester > Consero > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Steven Critchfield <critch@basesys.com>
tim.mcqueen@qualisys.biz
2003-Jun-29 20:10 UTC
[Asterisk-Users] Help! Problems talking to upstream switch
I'm new to *, but I've dealt with this issue on other switches. It sounds like either you are timing out getting DNIS information from your CO, or you are having trouble hearing the DTMF tones that are pulsed to you during the call setup process. Someone else on the list may know: is it possible to 1) configure the timeout for waiting on DNIS and 2) is it possible to change the Rx gain on the TDM cards? It's my understanding that the circuit is going throught the channel bank, which is acting like a drop-and-insert CSU by forwarding the 12 channels with voice to your * box. You mentioned that this was a Carrier Access channel bank, is it an ADIT 600? There are send and recieve gain settings on the channel bank unit that you might want to play with. HTH, HAND -Tim -----Original Message----- From: Andy Hester Sent: Sun 6/29/2003 8:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Help! Problems talking to upstream switch Hi, Please let me know if you have any ideas - I am taking wild guesses now.... Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP phones save a few fax machines. The customer has 7 digit unverified account codes on the trunks for billing purposes. The Problem: As I watch the console, I see calls coming in for exten "73" or "708" or "08" or "730" although most come in correctly (ie "7308"). My carrier has verified numerous times that they are sending 4 digits. I have 40 DID numbers that need to be routed and they are all in the 73xx range. I need to know anything that would cause my box randomly not to hear all 4 digits on occasion. Also, I have had trouble with people who dial out getting a congestion signal mainly on Long Distance numbers. The person would dial the number 4 or 5 times and get congestion then it might go through. Both of these conditions seem to be happening only about 10-20% of the time. What I have done: Moved a T100P card to its own IRQ to prevent problems with interrupts - Did not solve either issue. On the second try, got the carrier to change the way their switch chooses channels for incoming calls to prevent "glare" - This MAY have fixed the outgoing long distance issue as it seems to have gone away( although it doesn't seem logical to me that this would affect only LD) but did not fix incoming calls. Has anyone else had problems getting all of the digits that the Telco sends? Thanks, Andy Hester Consero _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 6530 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030629/7e7c5599/attachment.bin
Andy Hester
2003-Jun-29 21:27 UTC
[Asterisk-Users] Help! Problems talking to upstream switch
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