Voice mail isn't working on one extension.
I have the following connected to my FS box. A GrandStream phone, two POTS phones via the Linksys SPA2002, a soft phone installed on another system, not the FS machine and another POTS connected via an A200 port. If I call the GrandStream from one of the POTS on the SPA2002, it rings for ~30 seconds and I get a fast busy. If I call on the A200 connected phone, it rings and then nothing. It just stops ringing. If I call on the soft phone after it stops ringing I see a 404 error displayed on the phone "console". The log shows: 2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000@192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 404 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000@192.168.2.51 Standard HANGUP, cause: NO_ROUTE_DESTINATION If I do a diff of directory/default/1001.xml and 1002.xml the only differences are the numbers and passwords. Any idea what I might want to try next? Thanks, Jim.
On 08/26/2010 06:47 AM, Jim wrote:
I have the following connected to my FS box. A GrandStream phone, two POTS phones via the Linksys SPA2002, a soft phone installed on another system, not the FS machine and another POTS connected via an A200 port.
If I call the GrandStream from one of the POTS on the SPA2002, it rings for ~30 seconds and I get a fast busy. If I call on the A200 connected phone, it rings and then nothing. It just stops ringing. If I call on the soft phone after it stops ringing I see a 404 error displayed on the phone "console". The log shows:
2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000@192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 404 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000@192.168.2.51 Standard HANGUP, cause: NO_ROUTE_DESTINATION
If I do a diff of directory/default/1001.xml and 1002.xml the only differences are the numbers and passwords.
Any idea what I might want to try next?
Thanks, Jim.
I did change the extension from 1001 to 1010 just to see if it affected the results. It did not. Jim.
On 08/26/2010 09:48 AM, Jim wrote:
On 08/26/2010 06:47 AM, Jim wrote:
I have the following connected to my FS box. A GrandStream phone, two POTS phones via the Linksys SPA2002, a soft phone installed on another system, not the FS machine and another POTS connected via an A200 port.
If I call the GrandStream from one of the POTS on the SPA2002, it rings for ~30 seconds and I get a fast busy. If I call on the A200 connected phone, it rings and then nothing. It just stops ringing. If I call on the soft phone after it stops ringing I see a 404 error displayed on the phone "console". The log shows:
2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000@192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 404 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000@192.168.2.51 Standard HANGUP, cause: NO_ROUTE_DESTINATION
If I do a diff of directory/default/1001.xml and 1002.xml the only differences are the numbers and passwords.
Any idea what I might want to try next?
Thanks, Jim.
I did change the extension from 1001 to 1010 just to see if it affected the results. It did not.
Jim.
I also did a power off M/C of the phone and looked at the settings. They are mostly factory default and I know for a fact that voice mail worked with that phone when I was running Asterisk. I did a git of the latest yesterday to pickup freetdm so I'm running the latest and greatest code, or a very close likeness. I'm thinking voicemail on FS has some bugs. I also found that when I bridged an outside line (sip) to my A200 connected phone, the phone rings, lets me answer but if I let it ring, I never get voicemail. I do get VM if I dial it from a local extension. If I bridge to a POTS phone connected to the SPA2002 it works as expected, i. e. I get VM. Jim.
Can you send more of the debug from a call that works and one that fails? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jim Sent: Thursday, August 26, 2010 8:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voice mail isn't working on one extension. On 08/26/2010 06:47 AM, Jim wrote:
I have the following connected to my FS box. A GrandStream phone, two POTS phones via the Linksys SPA2002, a soft phone installed on another system, not the FS machine and another POTS connected via an A200 port.
If I call the GrandStream from one of the POTS on the SPA2002, it rings for ~30 seconds and I get a fast busy. If I call on the A200 connected phone, it rings and then nothing. It just stops ringing. If I call on the soft phone after it stops ringing I see a 404 error displayed on the phone "console". The log shows:
2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000@192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 404 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000@192.168.2.51 Standard HANGUP, cause: NO_ROUTE_DESTINATION
If I do a diff of directory/default/1001.xml and 1002.xml the only differences are the numbers and passwords.
Any idea what I might want to try next?
Thanks, Jim.
I did change the extension from 1001 to 1010 just to see if it affected the results. It did not. Jim. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org and then reply to this thread with the URL to your pastebin post. -MC On Thu, Aug 26, 2010 at 3:47 AM, Jim <jim@k4gvo.com> wrote:
I have the following connected to my FS box. A GrandStream phone, two POTS phones via the Linksys SPA2002, a soft phone installed on another system, not the FS machine and another POTS connected via an A200 port.
If I call the GrandStream from one of the POTS on the SPA2002, it rings for ~30 seconds and I get a fast busy. If I call on the A200 connected phone, it rings and then nothing. It just stops ringing. If I call on the soft phone after it stops ringing I see a 404 error displayed on the phone "console". The log shows:
2010-08-26 06:30:59.364714 [DEBUG] mod_sofia.c:453 Channel sofia/internal/1000@192.168.2.51 hanging up, cause: NO_ROUTE_DESTINATION 2010-08-26 06:30:59.384660 [DEBUG] mod_sofia.c:515 Responding to INVITE with: 404 2010-08-26 06:30:59.384660 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000@192.168.2.51 Standard HANGUP, cause: NO_ROUTE_DESTINATION
If I do a diff of directory/default/1001.xml and 1002.xml the only differences are the numbers and passwords.
Any idea what I might want to try next?
Thanks, Jim.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On 08/26/2010 12:00 PM, Michael Collins wrote:
Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org <http://pastebin.freeswitch.org> and then reply to this thread with the URL to your pastebin post. -MC
I don't like to waste computer space if it's not necessary. Sometimes the answer is simple and the answer doesn't require additional info. http://pastebin.freeswitch.org/13731 Note there are actually two problems demonstrated here. The first attempt is calling the GrandStream from a SPA2102 connected phone. The second is calling a POTS phone connected to the A200 from the SPA2102 and the third successful attempt is the reverse. There are notes at the beginning of the file giving approximate starting line numbers. Note, the second problem is intermittent. I though I had fixed it but I see I didn't. Thanks, Jim.
On 08/26/2010 12:31 PM, Jim wrote:
On 08/26/2010 12:00 PM, Michael Collins wrote:
Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org <http://pastebin.freeswitch.org> and then reply to this thread with the URL to your pastebin post. -MC
I don't like to waste computer space if it's not necessary. Sometimes the answer is simple and the answer doesn't require additional info.
http://pastebin.freeswitch.org/13731
Note there are actually two problems demonstrated here. The first attempt is calling the GrandStream from a SPA2102 connected phone. The second is calling a POTS phone connected to the A200 from the SPA2102 and the third successful attempt is the reverse. There are notes at the beginning of the file giving approximate starting line numbers.
Note, the second problem is intermittent. I though I had fixed it but I see I didn't.
Thanks, Jim. Not that it make a difference but I think that's a SPA2002.
Jim.
On 08/26/2010 12:50 PM, Jim wrote:
On 08/26/2010 12:31 PM, Jim wrote:
On 08/26/2010 12:00 PM, Michael Collins wrote:
Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org <http://pastebin.freeswitch.org> and then reply to this thread with the URL to your pastebin post. -MC
I don't like to waste computer space if it's not necessary. Sometimes the answer is simple and the answer doesn't require additional info.
http://pastebin.freeswitch.org/13731
Note there are actually two problems demonstrated here. The first attempt is calling the GrandStream from a SPA2102 connected phone. The second is calling a POTS phone connected to the A200 from the SPA2102 and the third successful attempt is the reverse. There are notes at the beginning of the file giving approximate starting line numbers.
Note, the second problem is intermittent. I though I had fixed it but I see I didn't.
Thanks, Jim. Not that it make a difference but I think that's a SPA2002.
Jim.
Looks like the line numbers changed from my vi session to pastebin. Ignore the notes at the top. Jim.
On Thu, Aug 26, 2010 at 9:31 AM, Jim <jim@k4gvo.com> wrote:
On 08/26/2010 12:00 PM, Michael Collins wrote:
Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org and then reply to this thread with the URL to your pastebin post. -MC
I don't like to waste computer space if it's not necessary. Sometimes the answer is simple and the answer doesn't require additional info.
http://pastebin.freeswitch.org/13731
Note there are actually two problems demonstrated here. The first attempt is calling the GrandStream from a SPA2102 connected phone. The second is calling a POTS phone connected to the A200 from the SPA2102 and the third successful attempt is the reverse. There are notes at the beginning of the file giving approximate starting line numbers.
#1 From what I can see, the GS is doing a redirect. Look at line 226 of the pastebin down to 240. Do you have call forwarding enabled on your GS? You may want to enable a SIP trace to see the exact SIP packet from the GS to see what it's doing. It is forwarding to destination number of "service" and there isn't an extension for that in your dialplan. (See lines leading up to 409 - "service" doesn't match anything until it gets to enum, where it eventually fails altogether at line 458.) #2 Does the phone connected to port 1 on the A200 even ring? It's interesting that there is a 9 second gap (between pb lines 732 and 733) which is where I would assume the ringing was taking place. -MC
On 08/26/2010 04:19 PM, Michael Collins wrote:
On Thu, Aug 26, 2010 at 9:31 AM, Jim <jim@k4gvo.com <mailto:jim@k4gvo.com>> wrote:
On 08/26/2010 12:00 PM, Michael Collins wrote:
Peder mentioned it already, but a complete log from start to finish would be helpful. Use pastebin.freeswitch.org <http://pastebin.freeswitch.org> and then reply to this thread with the URL to your pastebin post. -MC
I don't like to waste computer space if it's not necessary. Sometimes the answer is simple and the answer doesn't require additional info.
http://pastebin.freeswitch.org/13731
Note there are actually two problems demonstrated here. The first attempt is calling the GrandStream from a SPA2102 connected phone. The second is calling a POTS phone connected to the A200 from the SPA2102 and the third successful attempt is the reverse. There are notes at the beginning of the file giving approximate starting line numbers.
#1 From what I can see, the GS is doing a redirect. Look at line 226 of the pastebin down to 240. Do you have call forwarding enabled on your GS? You may want to enable a SIP trace to see the exact SIP packet from the GS to see what it's doing. It is forwarding to destination number of "service" and there isn't an extension for that in your dialplan. (See lines leading up to 409 - "service" doesn't match anything until it gets to enum, where it eventually fails altogether at line 458.)
How interesting. The phone is pretty simple, only a couple of pages and nothing that remotely mentions forwarding and/or "service". I guess I can put a service in the dial plan.
#2 Does the phone connected to port 1 on the A200 even ring? It's interesting that there is a 9 second gap (between pb lines 732 and 733) which is where I would assume the ringing was taking place.
Yes, it's ringing. I set the delay to 10 seconds rather than 30 when I was debugging it so I wouldn't have to wait so long. The phone rings, dials out, answers when I pick it up, but doesn't transfer to VM reliably. Thanks, Jim.
-MC
On 08/26/2010 04:19 PM, Michael Collins wrote:
#2 Does the phone connected to port 1 on the A200 even ring? It's interesting that there is a 9 second gap (between pb lines 732 and 733) which is where I would assume the ringing was taking place.
Looking at the log, I see where there was a EXECUTE ... answer followed immediately by a what looks like the state_core_machine telling FreeTDM to hang up, or am I reading that wrong? By the way, I do get a very short message before being dumped that says "Goodbye". She's pleasant, but curt. ;) By the way, I put a service in my dial plan and all is fine wrt the GS, thanks. Jim.
participants (3)
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Jim
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Michael Collins
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Peder