Re: Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
I'll be trying that as my next step but it seems that my other fresh - sipura 2000 unit that was sitting in the box which is running 1.0.15 firmware seems to work seamless so i find it odd ? that brand new unit works while the upgraded firmware ones don't? I'm not the only one having this same exact issue I've received 4 emails relating to the same issue from other users. So there's some kind trend going on with this? Unless someone else just happens to have gone through the exact same thing (probably not all that likely), it won't help to keep posting general statements. Really need to see something else like some output from debug, packet flows, etc. Since I don't have a sipura, I would not have a clue whether sip debug would provide anything, but I'd bet a few on this list might be able to interpret an etherial packet trace. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
Hello Rich, Sunday, January 25, 2004, 8:01:25 PM, you wrote: RA It would probably help if you used a packet sniffer (eg, ethereal) to look RA at the traffic, or at least provide the list with a useful clue other then RA it doesn't work. RA same here, when i recive an incoming call from x100p to line 1 on sipura, i can hear them but people can't hear me im using 1.0.24 on my firmware Miguel On Sun, 2004-01-25 at 20:54, Chris Higgins wrote: Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users RA ---End of Original Message- RA ___ RA Asterisk-Users mailing list RA [EMAIL PROTECTED] RA http://lists.digium.com/mailman/listinfo/asterisk-users RA To UNSUBSCRIBE or update options visit: RAhttp://lists.digium.com/mailman/listinfo/asterisk-users I'll be trying that as my next step but it seems that my other fresh - sipura 2000 unit that was sitting in the box which is running 1.0.15 firmware seems to work seamless so i find it odd ? that brand new unit works while the upgraded firmware ones don't? I'm not the only one having this same exact issue I've received 4 emails relating to the same issue from other users. So there's some kind trend going on with this? -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
Are you binding to one particular interface? I.e., Is bindaddr set to something besides 0.0.0.0? If so, ifdown your other interfaces, start asterisk, then bring up the other interfaces. If not, check your codecs. You need some lines like this in sip.conf: disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw allow=gsm If that still doesn't do it, I'll bet you're working some sort of NAT. Try the latest release of asterisk in the CVS and check the archives of this mail list for the new NAT parameters in sip.conf John - Original Message - From: Frankie Gravato [EMAIL PROTECTED] To: Asterisk [EMAIL PROTECTED] Sent: Sunday, January 25, 2004 12:22 AM Subject: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
I have a similar set up, I don't have a separate sip phone, but I have the same exact problem with the line 1. I don't know if my config files aren't right, but I can't transfer between exts yet, but my issues is with line one on an incoming call from an X100P as well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Higgins Sent: Sunday, January 25, 2004 3:54 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
same here, when i recive an incoming call from x100p to line 1 on sipura, i can hear them but people can't hear me im using 1.0.24 on my firmware Miguel On Sun, 2004-01-25 at 20:54, Chris Higgins wrote: Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
Steven E. Frazier wrote: I have a similar set up, I don't have a separate sip phone, but I have the same exact problem with the line 1. I don't know if my config files aren't right, but I can't transfer between exts yet, but my issues is with line one on an incoming call from an X100P as well. FYI.. I wasn't able to transfer between analog phones on my sipura until I explicitly set the DTMF mode on both lines in the sipura web configuration to INFO (which matched what I already had in sip.conf). -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
It would probably help if you used a packet sniffer (eg, ethereal) to look at the traffic, or at least provide the list with a useful clue other then it doesn't work. same here, when i recive an incoming call from x100p to line 1 on sipura, i can hear them but people can't hear me im using 1.0.24 on my firmware Miguel On Sun, 2004-01-25 at 20:54, Chris Higgins wrote: Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
