: Google voice incoming call doesn't work
Hi, I posted this issue last week. No one replied then the problem was gone after I recompiled and reinstalled everything. But from yesterday, it happened again. This time I couldn't get it work no matter how I tried. As I said, the outgoing works fine, only incoming doesn't. Now as a workaround I configured GV to forward calls to sipgate, then connect to sipgate as an SIP gateway to take the incoming calls. I am wondering whether anyone else got the same issue or I am the only unlucky one. Thanks! Jian On Thu, Nov 4, 2010 at 9:30 PM, Jian Ren <renjian@gmail.com> wrote:
mod_dingaling.c line 909:
if (!(tech_pvt->rtp_session <http://docs.freeswitch.org/structprivate__object.html#d5829b7e4a473ccebac90e44afe586c3> = switch_rtp_new(tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->ip <http://docs.freeswitch.org/structmdl__profile.html#38dfeac1fbf9e8d3784dd7eeb746cf6a>,00901 tech_pvt->local_port <http://docs.freeswitch.org/structprivate__object.html#63221941aab4357700497a529d51f2ca>,00902 tech_pvt->remote_ip <http://docs.freeswitch.org/structprivate__object.html#e1cbf100db6a0a4fd7eea10935fa2989>,00903 tech_pvt->remote_port <http://docs.freeswitch.org/structprivate__object.html#ca66c963cc0f4abf206ee9681d91a5ec>,00904 tech_pvt->codec_num <http://docs.freeswitch.org/structprivate__object.html#d1e23e3cec68cdff6135ef4c690aa8c4>,00905 tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->samples_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f6c2d7ee05d03910f7fb87cf3056f294>,00906 tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->microseconds_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f2bf6153bbb519fbb2523de02581c980>,00907 flags, tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->timer_name <http://docs.freeswitch.org/structmdl__profile.html#f34f270ada1c3c9c2203f4caba80eb8a>, &err, switch_core_session_get_pool(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>)))) {00908 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG <http://docs.freeswitch.org/switch__types_8h.html#bc899743e2a268974063fd82f654829f>(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>), SWITCH_LOG_DEBUG <http://docs.freeswitch.org/switch__types_8h.html#b74279dba7ee171b8b9d3decfcf13584a213e63b710064a30aa4524ddc30cbc4>, "RTP ERROR %s\n", err);00909 switch_channel_hangup <http://docs.freeswitch.org/group__switch__channel.html#g2d4ca1c1e6bfea7ba35665397b27ec60>(channel <http://docs.freeswitch.org/group__switch__channel.html#g991acd084cbac25bf6d2a325754ba221>, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER <http://docs.freeswitch.org/switch__types_8h.html#1606cd34d8539e5da76ebdcb89cc36b5f85efe65333a82f93a5e2913707575ae>);00910 return 0;
Something wrong with the rtp session, what's that?
Thanks!
On Thu, Nov 4, 2010 at 8:41 PM, Jian Ren <renjian@gmail.com> wrote:
Outcall works but the incoming doesn't. It always said "DESTINATION_OUT_OF_ORDER"
2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:784 New Channel dingaling/1001 [2d48674e-e874-11df-ab9a-9787aa669b3a] 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:782 Rename Channel dingaling/1001->DingaLing/new [2d48674e-e874-11df-ab9a-9787aa669b3a] 2010-11-04 20:32:31.797390 [NOTICE] mod_dingaling.c:1104 Ring-Ready DingaLing/new! 2010-11-04 20:32:32.164739 [NOTICE] mod_dingaling.c:909 Hangup DingaLing/new [CS_INIT] [DESTINATION_OUT_OF_ORDER] 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1242 Session 1 (DingaLing/new) Ended 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1244 Close Channel DingaLing/new [CS_DESTROY]
I had skypopen loaded but pointed to another extension, not sure whether it's related. Now I commented out mod_skypopen but still got the same problem. Please help!
My jingling client.xml:
<include> <!-- Client Profile (Original mode) --> <!-- to use this profile take the x- away from the open and close tags so its <profile> and$ <profile type="client"> <param name="name" value="$${xmpp_client_profile}"/> <param name="login" value="xxxx@gmail.com/gtalk"/> <param name="password" value="xxxx"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <!-- <param name="ext-rtp-ip" value="auto-nat"/> --> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="plain"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <param name="server" value="talk.google.com"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="1001"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <param name="vad" value="both"/> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks! Jian
hi for incoming call for me mostly no problem, outgoing sometimes failed and in the next time works normally. when outgoing call problem happens, i try from gmail as well and from gmail it works. so i have no idea about this issue thx budi On Fri, Nov 12, 2010 at 4:08 AM, Jian Ren <renjian@gmail.com> wrote:
Hi, I posted this issue last week. No one replied then the problem was gone after I recompiled and reinstalled everything. But from yesterday, it happened again. This time I couldn't get it work no matter how I tried. As I said, the outgoing works fine, only incoming doesn't. Now as a workaround I configured GV to forward calls to sipgate, then connect to sipgate as an SIP gateway to take the incoming calls. I am wondering whether anyone else got the same issue or I am the only unlucky one.
Thanks! Jian
On Thu, Nov 4, 2010 at 9:30 PM, Jian Ren <renjian@gmail.com> wrote:
mod_dingaling.c line 909:
if (!(tech_pvt->rtp_session <http://docs.freeswitch.org/structprivate__object.html#d5829b7e4a473ccebac90e44afe586c3> = switch_rtp_new(tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->ip <http://docs.freeswitch.org/structmdl__profile.html#38dfeac1fbf9e8d3784dd7eeb746cf6a>,00901 tech_pvt->local_port <http://docs.freeswitch.org/structprivate__object.html#63221941aab4357700497a529d51f2ca>,00902 tech_pvt->remote_ip <http://docs.freeswitch.org/structprivate__object.html#e1cbf100db6a0a4fd7eea10935fa2989>,00903 tech_pvt->remote_port <http://docs.freeswitch.org/structprivate__object.html#ca66c963cc0f4abf206ee9681d91a5ec>,00904 tech_pvt->codec_num <http://docs.freeswitch.org/structprivate__object.html#d1e23e3cec68cdff6135ef4c690aa8c4>,00905 tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->samples_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f6c2d7ee05d03910f7fb87cf3056f294>,00906 tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->microseconds_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f2bf6153bbb519fbb2523de02581c980>,00907 flags, tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->timer_name <http://docs.freeswitch.org/structmdl__profile.html#f34f270ada1c3c9c2203f4caba80eb8a>, &err, switch_core_session_get_pool(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>)))) {00908 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG <http://docs.freeswitch.org/switch__types_8h.html#bc899743e2a268974063fd82f654829f>(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>), SWITCH_LOG_DEBUG <http://docs.freeswitch.org/switch__types_8h.html#b74279dba7ee171b8b9d3decfcf13584a213e63b710064a30aa4524ddc30cbc4>, "RTP ERROR %s\n", err);00909 switch_channel_hangup <http://docs.freeswitch.org/group__switch__channel.html#g2d4ca1c1e6bfea7ba35665397b27ec60>(channel <http://docs.freeswitch.org/group__switch__channel.html#g991acd084cbac25bf6d2a325754ba221>, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER <http://docs.freeswitch.org/switch__types_8h.html#1606cd34d8539e5da76ebdcb89cc36b5f85efe65333a82f93a5e2913707575ae>);00910 return 0;
Something wrong with the rtp session, what's that?
Thanks!
On Thu, Nov 4, 2010 at 8:41 PM, Jian Ren <renjian@gmail.com> wrote:
Outcall works but the incoming doesn't. It always said "DESTINATION_OUT_OF_ORDER"
2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:784 New Channel dingaling/1001 [2d48674e-e874-11df-ab9a-9787aa669b3a] 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:782 Rename Channel dingaling/1001->DingaLing/new [2d48674e-e874-11df-ab9a-9787aa669b3a] 2010-11-04 20:32:31.797390 [NOTICE] mod_dingaling.c:1104 Ring-Ready DingaLing/new! 2010-11-04 20:32:32.164739 [NOTICE] mod_dingaling.c:909 Hangup DingaLing/new [CS_INIT] [DESTINATION_OUT_OF_ORDER] 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1242 Session 1 (DingaLing/new) Ended 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1244 Close Channel DingaLing/new [CS_DESTROY]
I had skypopen loaded but pointed to another extension, not sure whether it's related. Now I commented out mod_skypopen but still got the same problem. Please help!
My jingling client.xml:
<include> <!-- Client Profile (Original mode) --> <!-- to use this profile take the x- away from the open and close tags so its <profile> and$ <profile type="client"> <param name="name" value="$${xmpp_client_profile}"/> <param name="login" value="xxxx@gmail.com/gtalk"/> <param name="password" value="xxxx"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <!-- <param name="ext-rtp-ip" value="auto-nat"/> --> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="plain"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <param name="server" value="talk.google.com"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="1001"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <param name="vad" value="both"/> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks! Jian
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Why is it that on the month of November since 2007, somebody keeps asking question about FS to GV and vice-versa? And guess what, the same old question which nobody damn care to answer. To make the record straight, this kind of setup has been working on FS since the summer of 2007. It just take some patience and testing on how to make it work. I am not sure if you are configuring FS jingle/dingaling module to be the client for GV or you have FS configured with a Jabber server and this Jabber server does a S2S with the google Jabber server. Anyway...I have this kind of setup working for ages already.... :q! DIB
Hi, I posted this issue last week. No one replied then the problem was gone after I recompiled and reinstalled everything. But from yesterday, it happened again. This time I couldn't get it work no matter how I tried. As I said, the outgoing works fine, only incoming doesn't. Now as a workaround I configured GV to forward calls to sipgate, then connect to sipgate as an SIP gateway to take the incoming calls. I am wondering whether anyone else got the same issue or I am the only unlucky one.
Thanks! Jian
On Thu, Nov 4, 2010 at 9:30 PM, Jian Ren <renjian@gmail.com> wrote:
mod_dingaling.c line 909:
if (!(tech_pvt->rtp_session <http://docs.freeswitch.org/structprivate__object.html#d5829b7e4a473ccebac90e44afe586c3> = switch_rtp_new(tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->ip <http://docs.freeswitch.org/structmdl__profile.html#38dfeac1fbf9e8d3784dd7eeb746cf6a>,00901
tech_pvt->local_port <http://docs.freeswitch.org/structprivate__object.html#63221941aab4357700497a529d51f2ca>,00902
tech_pvt->remote_ip <http://docs.freeswitch.org/structprivate__object.html#e1cbf100db6a0a4fd7eea10935fa2989>,00903
tech_pvt->remote_port <http://docs.freeswitch.org/structprivate__object.html#ca66c963cc0f4abf206ee9681d91a5ec>,00904
tech_pvt->codec_num <http://docs.freeswitch.org/structprivate__object.html#d1e23e3cec68cdff6135ef4c690aa8c4>,00905
tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->samples_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f6c2d7ee05d03910f7fb87cf3056f294>,00906
tech_pvt->read_codec <http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation <http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->microseconds_per_packet <http://docs.freeswitch.org/structswitch__codec__implementation.html#f2bf6153bbb519fbb2523de02581c980>,00907
flags, tech_pvt->profile <http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->timer_name <http://docs.freeswitch.org/structmdl__profile.html#f34f270ada1c3c9c2203f4caba80eb8a>, &err, switch_core_session_get_pool(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>)))) {00908 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG <http://docs.freeswitch.org/switch__types_8h.html#bc899743e2a268974063fd82f654829f>(tech_pvt->session <http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>), SWITCH_LOG_DEBUG <http://docs.freeswitch.org/switch__types_8h.html#b74279dba7ee171b8b9d3decfcf13584a213e63b710064a30aa4524ddc30cbc4>, "RTP ERROR %s\n", err);00909 switch_channel_hangup <http://docs.freeswitch.org/group__switch__channel.html#g2d4ca1c1e6bfea7ba35665397b27ec60>(channel <http://docs.freeswitch.org/group__switch__channel.html#g991acd084cbac25bf6d2a325754ba221>, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER <http://docs.freeswitch.org/switch__types_8h.html#1606cd34d8539e5da76ebdcb89cc36b5f85efe65333a82f93a5e2913707575ae>);00910 return 0;
Something wrong with the rtp session, what's that?
Thanks!
On Thu, Nov 4, 2010 at 8:41 PM, Jian Ren <renjian@gmail.com> wrote:
Outcall works but the incoming doesn't. It always said "DESTINATION_OUT_OF_ORDER"
2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:784 New Channel dingaling/1001 [2d48674e-e874-11df-ab9a-9787aa669b3a] 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:782 Rename Channel dingaling/1001->DingaLing/new [2d48674e-e874-11df-ab9a-9787aa669b3a] 2010-11-04 20:32:31.797390 [NOTICE] mod_dingaling.c:1104 Ring-Ready DingaLing/new! 2010-11-04 20:32:32.164739 [NOTICE] mod_dingaling.c:909 Hangup DingaLing/new [CS_INIT] [DESTINATION_OUT_OF_ORDER] 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1242 Session 1 (DingaLing/new) Ended 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1244 Close Channel DingaLing/new [CS_DESTROY]
I had skypopen loaded but pointed to another extension, not sure whether it's related. Now I commented out mod_skypopen but still got the same problem. Please help!
My jingling client.xml:
<include> <!-- Client Profile (Original mode) --> <!-- to use this profile take the x- away from the open and close tags so its <profile> and$ <profile type="client"> <param name="name" value="$${xmpp_client_profile}"/> <param name="login" value="xxxx@gmail.com/gtalk"/> <param name="password" value="xxxx"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <!-- <param name="ext-rtp-ip" value="auto-nat"/> --> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="plain"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <param name="server" value="talk.google.com"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="1001"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <param name="vad" value="both"/> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks! Jian
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demuel wrote:
Why is it that on the month of November since 2007, somebody keeps asking question about FS to GV and vice-versa? And guess what, the same old question which nobody damn care to answer. Exactly. Just a newbie thought s/he could get help from the list, it turns out experts here could careless to help.
To make the record straight, this kind of setup has been working on FS
since the summer of 2007. It just take some patience and testing on how to make it work. The new Google Voice through GMail by means of GTalk has just been recently (09/2010?) added to the trunk. So, what you were saying above is a complete different from the OP is asking.
Anyway...I have this kind of setup working for ages already.... Why don't you post your complete configuration w.r.t to this once and for all. This way newbies with such a question will be able to search through the list before posting asking the same question again and again. Better off, update the Wiki to include a complete configuration with dialplans to handle GTalk calls. IIRC, the existing Wiki lacks this. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-voice-incoming-call-doe... Sent from the freeswitch-users mailing list archive at Nabble.com.
The problem is, it doesn't always fail. I did another re-installation and now it's working again. Since no one cares so far, I guess I would just ignore this for now, and will setup sipgate too as a backup, the chance both fail might be very low. Thanks! Jian On Fri, Nov 12, 2010 at 10:26 AM, mazilo <Nabble@slickdeals.endjunk.com>wrote:
demuel wrote:
Why is it that on the month of November since 2007, somebody keeps asking question about FS to GV and vice-versa? And guess what, the same old question which nobody damn care to answer. Exactly. Just a newbie thought s/he could get help from the list, it turns out experts here could careless to help.
To make the record straight, this kind of setup has been working on FS
since the summer of 2007. It just take some patience and testing on how to make it work. The new Google Voice through GMail by means of GTalk has just been recently (09/2010?) added to the trunk. So, what you were saying above is a complete different from the OP is asking.
Anyway...I have this kind of setup working for ages already.... Why don't you post your complete configuration w.r.t to this once and for all. This way newbies with such a question will be able to search through the list before posting asking the same question again and again. Better off, update the Wiki to include a complete configuration with dialplans to handle GTalk calls. IIRC, the existing Wiki lacks this.
----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-voice-incoming-call-doe... Sent from the freeswitch-users mailing list archive at Nabble.com.
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participants (4)
-
budi wibowo
-
demuel＠thephinix.org
-
Jian Ren
-
mazilo