Re: [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)
I just wanted to follow up with the solution for the searchable archives. To summarize: Adtran TA 750 w/ FXO lines from a Nortel Meridien Problem: DTMF digits not being decoded properly Solution: Set the TX/RX Attenuation on the TA750 to 9.0db (max) The sound is a little low now, but DTMF is working properly. -gn On Dec 9, 2005, at 12:56 PM, Gaurav Naik wrote: Update: I've determined that the problem is DTMF 9. I cannot get to extension 6950 on the Nortel. The 9 is totally skipped. However, if I dial 69501, I get connected to extension 6501. What is so special about DTMF 9? Thanks On Dec 8, 2005, at 7:40 PM, Gaurav Naik wrote: I'm having a strange problem with an analog line connected to an Adtran Channel Bank. It seems as tho, I cannot make outgoing calls out of the PBX the analog lines are connected to. I'll explain... The channel bank has a few analog lines (loop start) coming in to the FXO cards from a Nortel Meridien Option 81c. I have no administrative/technical control over the Nortel, so I have to believe that the lines are configured correctly (nor are they willing to setup a PRI). As a start, I terminated one of the analog lines into an analog phone and was able to successfully make and receive calls. Next, I connected this same line to a X100P and had no problems with Asterisk (other than disconnect supervision). Finally, I connected the line to the channel bank (channel 1 of the T1). The T1 between TE110P and the Channel Bank is up -- no errors. The TE110P is the master, with the 750 as the slave. I could dial a 4-digit extension number and connect to any station on the Nortel switch (albeit with some echo, but no static). However, dialing outside of the Nortel doesn't work. A '9' has to be dialed in order to get an outside line. First, I started with a SIP phone, and setup my dial plan accordingly. Stuck a few No-Ops in there and watched on the console. Asterisk was correctly dialing 91800 on the FXO port, however, the Nortel kept ringing extension 1800. Next, I tried adding waits (w9w1800XXX), but I still keep getting extension 1800. I even tried longer and longer waits. I turned on zap debugging from the asterisk CLI, and could see that DTMF 9 was being sent. Hmm. Again, with X100P, I could dial outside no problem -- using the SAME dial plan. In order to make debugging this problem easier, I hooked up the analog phone to one of the FXS ports, and had Asterisk do a native bridge of the two ZAP lines. Fine, so now I'm hearing the dialtone from the Nortel on my analog set. I dial '9', there a slight silence (500ms or so) and I get a dial tone. Correct behavior. Next, I dial 1800extension 1800 is ringing. That didn't work. I cycled thru about 20 different configurations of zapata.conf and zaptel.conf. Still doesn't work. I'm beginning to wonder if there is something wrong with the Adtran. For a change, I decided to try incoming calls. Asterisk gets the ring and answers (although it does report something about the line ringing in the wrong state), however, it cannot decode the DTMF digits the caller is dialing. I dial 813 and Asterisk thinks I dialed 83. My dial plan to setup to report an invalid extension and gives the user another chance to enter the extension. The second try always works. (I did this about 5 times, with no problem.) Hmm. So it gets the wrong DTMF digits the first time, every time. I checked the Digit/Response timeouts...they are set to 5 (for both cases). I even tried the relaxdtmf directive, but to know avail. Again, I wasn't having these problems on X100P. The only difference is that the X100P is running Asterisk v1.0.9 and is in an older machine. I've gone through every configuration directive possible (disabling usecallerid, callprogress, echocancel, etc. etc. ). I've even tried dialing 9 three times. Then I pulled out a multi- meter and made sure that Tip and Ring weren't reversed (although that shouldn't make a difference). I checked zttool for IRQ misses and it reported none. The dialtone sounds fine (no hiss, pops, or static), but after checking the asterisk/zaptel configuration 25 times, I'm beginning to think its a wiring problem. Its possible that the 750 isn't grounded correctly..that is probably my next step. The relevant portions of my current configuration follow. Any assistance, or hints and tips for debugging this problem are appreciated. Thanks in advance, -- Gaurav Naik ...apologizing for the long e-mail. *** System Config -- Dell Poweredge 1550 server Digium TE110P (master clock) Adtran TA 750 (slave, and two 4-port FXO cards, and 1 4-port FXS card) Asterisk/Zapata v1.2 RHEL v4.0 Advanced Server Polycom/Grandstream SIP Phones a few plain old analog phones /etc/zaptel.conf -- span=1,0,0,esf,b8zs fxsls=1 fxols=9 loadzone = us defaultzone=us
Re: [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)
Update: I've determined that the problem is DTMF 9. I cannot get to extension 6950 on the Nortel. The 9 is totally skipped. However, if I dial 69501, I get connected to extension 6501. What is so special about DTMF 9? Thanks On Dec 8, 2005, at 7:40 PM, Gaurav Naik wrote: I'm having a strange problem with an analog line connected to an Adtran Channel Bank. It seems as tho, I cannot make outgoing calls out of the PBX the analog lines are connected to. I'll explain... The channel bank has a few analog lines (loop start) coming in to the FXO cards from a Nortel Meridien Option 81c. I have no administrative/technical control over the Nortel, so I have to believe that the lines are configured correctly (nor are they willing to setup a PRI). As a start, I terminated one of the analog lines into an analog phone and was able to successfully make and receive calls. Next, I connected this same line to a X100P and had no problems with Asterisk (other than disconnect supervision). Finally, I connected the line to the channel bank (channel 1 of the T1). The T1 between TE110P and the Channel Bank is up -- no errors. The TE110P is the master, with the 750 as the slave. I could dial a 4-digit extension number and connect to any station on the Nortel switch (albeit with some echo, but no static). However, dialing outside of the Nortel doesn't work. A '9' has to be dialed in order to get an outside line. First, I started with a SIP phone, and setup my dial plan accordingly. Stuck a few No-Ops in there and watched on the console. Asterisk was correctly dialing 91800 on the FXO port, however, the Nortel kept ringing extension 1800. Next, I tried adding waits (w9w1800XXX), but I still keep getting extension 1800. I even tried longer and longer waits. I turned on zap debugging from the asterisk CLI, and could see that DTMF 9 was being sent. Hmm. Again, with X100P, I could dial outside no problem -- using the SAME dial plan. In order to make debugging this problem easier, I hooked up the analog phone to one of the FXS ports, and had Asterisk do a native bridge of the two ZAP lines. Fine, so now I'm hearing the dialtone from the Nortel on my analog set. I dial '9', there a slight silence (500ms or so) and I get a dial tone. Correct behavior. Next, I dial 1800extension 1800 is ringing. That didn't work. I cycled thru about 20 different configurations of zapata.conf and zaptel.conf. Still doesn't work. I'm beginning to wonder if there is something wrong with the Adtran. For a change, I decided to try incoming calls. Asterisk gets the ring and answers (although it does report something about the line ringing in the wrong state), however, it cannot decode the DTMF digits the caller is dialing. I dial 813 and Asterisk thinks I dialed 83. My dial plan to setup to report an invalid extension and gives the user another chance to enter the extension. The second try always works. (I did this about 5 times, with no problem.) Hmm. So it gets the wrong DTMF digits the first time, every time. I checked the Digit/Response timeouts...they are set to 5 (for both cases). I even tried the relaxdtmf directive, but to know avail. Again, I wasn't having these problems on X100P. The only difference is that the X100P is running Asterisk v1.0.9 and is in an older machine. I've gone through every configuration directive possible (disabling usecallerid, callprogress, echocancel, etc. etc. ). I've even tried dialing 9 three times. Then I pulled out a multi-meter and made sure that Tip and Ring weren't reversed (although that shouldn't make a difference). I checked zttool for IRQ misses and it reported none. The dialtone sounds fine (no hiss, pops, or static), but after checking the asterisk/zaptel configuration 25 times, I'm beginning to think its a wiring problem. Its possible that the 750 isn't grounded correctly..that is probably my next step. The relevant portions of my current configuration follow. Any assistance, or hints and tips for debugging this problem are appreciated. Thanks in advance, -- Gaurav Naik ...apologizing for the long e-mail. *** System Config -- Dell Poweredge 1550 server Digium TE110P (master clock) Adtran TA 750 (slave, and two 4-port FXO cards, and 1 4-port FXS card) Asterisk/Zapata v1.2 RHEL v4.0 Advanced Server Polycom/Grandstream SIP Phones a few plain old analog phones /etc/zaptel.conf -- span=1,0,0,esf,b8zs fxsls=1 fxols=9 loadzone = us defaultzone=us /etc/asterisk/zapata.conf (this is version 34325, the simplest one). -- signalling=fxo_ls language=en context=from-analog-phone channel = 9 signalling=fxs_ls language=en context=from-outside channel = 1 Adtran TA 750 -- FXO Loop Start (Time Slot 1) TX Attenuation 0.0 RX Attenuation 0.0 All other FXO ports are disabled. (i've tried it with all the FXO ports enabled as
[Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)
I'm having a strange problem with an analog line connected to an Adtran Channel Bank. It seems as tho, I cannot make outgoing calls out of the PBX the analog lines are connected to. I'll explain... The channel bank has a few analog lines (loop start) coming in to the FXO cards from a Nortel Meridien Option 81c. I have no administrative/technical control over the Nortel, so I have to believe that the lines are configured correctly (nor are they willing to setup a PRI). As a start, I terminated one of the analog lines into an analog phone and was able to successfully make and receive calls. Next, I connected this same line to a X100P and had no problems with Asterisk (other than disconnect supervision). Finally, I connected the line to the channel bank (channel 1 of the T1). The T1 between TE110P and the Channel Bank is up -- no errors. The TE110P is the master, with the 750 as the slave. I could dial a 4-digit extension number and connect to any station on the Nortel switch (albeit with some echo, but no static). However, dialing outside of the Nortel doesn't work. A '9' has to be dialed in order to get an outside line. First, I started with a SIP phone, and setup my dial plan accordingly. Stuck a few No-Ops in there and watched on the console. Asterisk was correctly dialing 91800 on the FXO port, however, the Nortel kept ringing extension 1800. Next, I tried adding waits (w9w1800XXX), but I still keep getting extension 1800. I even tried longer and longer waits. I turned on zap debugging from the asterisk CLI, and could see that DTMF 9 was being sent. Hmm. Again, with X100P, I could dial outside no problem -- using the SAME dial plan. In order to make debugging this problem easier, I hooked up the analog phone to one of the FXS ports, and had Asterisk do a native bridge of the two ZAP lines. Fine, so now I'm hearing the dialtone from the Nortel on my analog set. I dial '9', there a slight silence (500ms or so) and I get a dial tone. Correct behavior. Next, I dial 1800extension 1800 is ringing. That didn't work. I cycled thru about 20 different configurations of zapata.conf and zaptel.conf. Still doesn't work. I'm beginning to wonder if there is something wrong with the Adtran. For a change, I decided to try incoming calls. Asterisk gets the ring and answers (although it does report something about the line ringing in the wrong state), however, it cannot decode the DTMF digits the caller is dialing. I dial 813 and Asterisk thinks I dialed 83. My dial plan to setup to report an invalid extension and gives the user another chance to enter the extension. The second try always works. (I did this about 5 times, with no problem.) Hmm. So it gets the wrong DTMF digits the first time, every time. I checked the Digit/Response timeouts...they are set to 5 (for both cases). I even tried the relaxdtmf directive, but to know avail. Again, I wasn't having these problems on X100P. The only difference is that the X100P is running Asterisk v1.0.9 and is in an older machine. I've gone through every configuration directive possible (disabling usecallerid, callprogress, echocancel, etc. etc. ). I've even tried dialing 9 three times. Then I pulled out a multi-meter and made sure that Tip and Ring weren't reversed (although that shouldn't make a difference). I checked zttool for IRQ misses and it reported none. The dialtone sounds fine (no hiss, pops, or static), but after checking the asterisk/zaptel configuration 25 times, I'm beginning to think its a wiring problem. Its possible that the 750 isn't grounded correctly..that is probably my next step. The relevant portions of my current configuration follow. Any assistance, or hints and tips for debugging this problem are appreciated. Thanks in advance, -- Gaurav Naik ...apologizing for the long e-mail. *** System Config -- Dell Poweredge 1550 server Digium TE110P (master clock) Adtran TA 750 (slave, and two 4-port FXO cards, and 1 4-port FXS card) Asterisk/Zapata v1.2 RHEL v4.0 Advanced Server Polycom/Grandstream SIP Phones a few plain old analog phones /etc/zaptel.conf -- span=1,0,0,esf,b8zs fxsls=1 fxols=9 loadzone = us defaultzone=us /etc/asterisk/zapata.conf (this is version 34325, the simplest one). -- signalling=fxo_ls language=en context=from-analog-phone channel = 9 signalling=fxs_ls language=en context=from-outside channel = 1 Adtran TA 750 -- FXO Loop Start (Time Slot 1) TX Attenuation 0.0 RX Attenuation 0.0 All other FXO ports are disabled. (i've tried it with all the FXO ports enabled as well). FXS Loop Start (Time Slot 9) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
