Hello, I am attempting to get Gafachi setup on our system. They are telling me that the caller id is not being sent with the request, and are thus rejecting the calls. Is there a short FAQ on how to send the caller id? in vars.xml, I have: <X-PRE-PROCESS cmd="set" data="outbound_caller_name=My Name"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=XXXXXXXXXX"/> call.js: call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX"; sessionx = new Session(); rs = sessionx.originate(session, call, timeout); I've also tried setting various forms of {caller_id_name}, {origination_caller_id_name}, {effective_caller_id_name} / number, but with little success. I am unsure which of the above to send in this context. Where in the trace / logs / session variables can I / should I see the caller_id being sent? Is there a setting I'm missing someplace? I'm sure I've missed something simple. thanks for your help, Nick
On Jun 11, 2008, at 8:18 PM, Nick Temple wrote:
call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
Try to export instead of set. The variable isn't on the b-leg so you actually have it set on the a-leg or you can try this which should be documented rather well on the wiki: call = "{effective_caller_id_number =12345,ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX"; /b
I see where the caller_id_number is being set in: switch_core_session_outgoing_channel However my tests are showing that It looks like in , this test is returning false (no session): if (session) { .... } hence it appears that the caller_id_number part of the struct is not being set from th effective_caller_id_number. Later in the code it's looking for a channel if (channel) { .... }, which is also null, making the entire function pretty much a no-op in this case. Am I even looking in the right place? Thanks, Nick On Wed, Jun 11, 2008 at 9:28 PM, Brian West <brian@freeswitch.org> wrote:
On Jun 11, 2008, at 8:18 PM, Nick Temple wrote:
call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
Try to export instead of set. The variable isn't on the b-leg so you actually have it set on the a-leg or you can try this which should be documented rather well on the wiki:
call = "{effective_caller_id_number=12345,ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
/b
_______________________________________________ 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
I don't get what you're doing here.. effective_caller_id_number and origination_caller_id_number depending on if you're originating or if you're doing a bridge. I have verified they work. Can you elaborate on your situation because you shouldn't have to go digging in the code for this. /b On Jun 12, 2008, at 2:10 AM, Nick Temple wrote:
I see where the caller_id_number is being set in: switch_core_session_outgoing_channel
However my tests are showing that
It looks like in , this test is returning false (no session):
if (session) { .... }
hence it appears that the caller_id_number part of the struct is not being set from th effective_caller_id_number. Later in the code it's looking for a channel if (channel) { .... }, which is also null, making the entire function pretty much a no-op in this case.
Am I even looking in the right place?
Thanks,
Nick
On Wed, Jun 11, 2008 at 9:28 PM, Brian West <brian@freeswitch.org> wrote:
On Jun 11, 2008, at 8:18 PM, Nick Temple wrote:
call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
Try to export instead of set. The variable isn't on the b-leg so you actually have it set on the a-leg or you can try this which should be documented rather well on the wiki:
call = "{effective_caller_id_number =12345,ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
/b
_______________________________________________ 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
_______________________________________________ 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
Brian West sip:brian@freeswitch.org
Using Javascript, I'm simply trying to originate a call: (variables sent over XMLRPC from something similar to: jsrun alert_cid.js $id $wave $phone $gateway $timeout $caller_id_number $caller_id_name ) console_log('notice', trace(session, "GOT: (" + callid + "," + greeting + "," + phone + "," + gw + ","+ timeout + "," + caller_id + "," + caller_name + ")\n")); sessionx = new Session(); var call = "{origination_caller_id_number=XXXXXXXX,origination_caller_id_name=XXXX,ignore_early_media=true," + "originate_timeout=" + timeout + "}" + "sofia/gateway/" + gw + "/" + phone; console_log('notice', trace(session, "originate: " + call + "\n")); console_log('notice', session.caller_id_Name + "\n"); rs = sessionx.originate(session, call, timeout); console_log('debug', trace(session, "origination complete: " + rs + "\n")); console_log('notice', sessionx.caller_id_Name + "\n"); (trace is an internal function that just adds some additional information, such as cause, to the string) I've traced the SIP calls, so has Gafachi, and we're not seeing the caller_id. They are rejecting the calls because of this: SIP/2.0 604 Invalid Calling Party Number Via: SIP/2.0/UDP 75.101.133.137:5080 ;branch=z9hG4bKm4028av6cyD0F;received=75.101.133.137;rport=5080 From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com<sip%3Aa6031sTVQLeOhyIV@sip.gafachi.com> ;transport=udp>;tag=j8BBv5jaycaFF To: <sip:15025482003@sip.gafachi.com <sip%3A15025482003@sip.gafachi.com>
;tag=gss0c342e9c3ad2398e2f610f7ce613a089 Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Server: Gafachi UAS v110.08 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: <sip:15025482003@64.192.112.13 <sip%3A15025482003@64.192.112.13>
Content-Length: 0
From reading various docs, I assume that I'm looking for the caller_id_number in the FROM header of the INVITE sent to the gateway?
Here's my internal logs after running the code, above: 2008-06-12 12:08:52 [NOTICE] alert_cid.js:1 console_log() =>[undefined] (undefined) GOT: (54,/tmp/4dc180ba-37fd-1e14-ad0c-7aa0770f2d0e.wav,XX,default,30,XX,XX) 2008-06-12 12:08:52 [NOTICE] alert_cid.js:1 console_log() =>[undefined] (undefined) originate: {origination_caller_id_number=XX,origination_caller_id_name=XX,ignore_early_media=true,originate_timeout=30}sofia/gateway/default/XX 2008-06-12 12:08:52 [NOTICE] alert_cid.js:1 console_log() undefined <=== caller_id_number should be defined here for sure 2008-06-12 12:08:52 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/external/XX [d1be5806-7535-4168-8ad2-e6d787f603bd] [SIP TRACE DELETED. relevent error portion quoted above] 2008-06-12 12:08:53 [NOTICE] sofia.c:2085 sofia_handle_sip_i_state() Hangup sofia/external/1502XXXXXXX [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2008-06-12 12:08:53 [WARNING] mod_spidermonkey.c:2771 session_originate() Cannot Create Outgoing Channel! [{origination_caller_id_number=502XXXXXXX,origination_caller_id_name=XXXX,ignore_early_media=true,originate_timeout=30}sofia/gateway/default/1502XXXXXXX] Obviously from the note below effective_caller_id_name was not correct, I should have been using origination_caller_id_xxxx instead ... I've tried both many times, I just happened to start tracing (grepping) effective_caller_id_name first to see if I could get a handle on what was going on. I just know it's something simple I've missed ... at this point I'm just going through the code and adding additional logging to the routines so I can try to figure out what it is I've missed, mis-spelled or whatever - or even hard-code the caller_id if necessary to get past this (seemingly tiny) hurdle. I'm at work right now, I'll try to get on IRC this evening or tomorrow. Nick On Thu, Jun 12, 2008 at 9:20 AM, Brian West <brian@freeswitch.org> wrote:
I don't get what you're doing here.. effective_caller_id_number and origination_caller_id_number depending on if you're originating or if you're doing a bridge. I have verified they work. Can you elaborate on your situation because you shouldn't have to go digging in the code for this. /b
On Jun 12, 2008, at 2:10 AM, Nick Temple wrote:
I see where the caller_id_number is being set in: switch_core_session_outgoing_channel
However my tests are showing that
It looks like in , this test is returning false (no session):
if (session) { .... }
hence it appears that the caller_id_number part of the struct is not being set from th effective_caller_id_number. Later in the code it's looking for a channel if (channel) { .... }, which is also null, making the entire function pretty much a no-op in this case.
Am I even looking in the right place?
Thanks,
Nick
On Wed, Jun 11, 2008 at 9:28 PM, Brian West <brian@freeswitch.org> wrote:
On Jun 11, 2008, at 8:18 PM, Nick Temple wrote:
call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
Try to export instead of set. The variable isn't on the b-leg so you actually have it set on the a-leg or you can try this which should be documented rather well on the wiki:
call = "{effective_caller_id_number=12345,ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
/b
_______________________________________________ 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
_______________________________________________ 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
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
Show me the INVITE. On Jun 12, 2008, at 11:50 AM, Nick Temple wrote:
SIP/2.0 604 Invalid Calling Party Number Via: SIP/2.0/UDP 75.101.133.137 :5080;branch=z9hG4bKm4028av6cyD0F;received=75.101.133.137;rport=5080 From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com;transport=udp>;tag=j8BBv5jaycaFF To: <sip: 15025482003@sip.gafachi.com>;tag=gss0c342e9c3ad2398e2f610f7ce613a089 Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Server: Gafachi UAS v110.08 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: <sip:15025482003@64.192.112.13> Content-Length: 0
Brian West sip:brian@freeswitch.org
Show me the INVITE: INVITE sip:1502XXXXXXX@sip.gafachi.com<sip%3A1502XXXXXXX@sip.gafachi.com>SIP/2.0 Via: SIP/2.0/UDP 75.101.133.137:5080;rport;branch=z9hG4bKm4028av6cyD0F Max-Forwards: 70 From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com<sip%3Aa6031sTVQLeOhyIV@sip.gafachi.com> ;transport=udp>;tag=j8BBv5jaycaFF To: <sip:15025482003@sip.gafachi.com <sip%3A15025482003@sip.gafachi.com>> Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Contact: <sip:a6031sTVQLeOhyIV@75.101.133.137:5080>;transport=udp Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.0-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Allow-Events: talk Proxy-Authorization: Digest username="a6031sTVQLeOhyIV", realm=" sip.gafachi.com", nonce="4fa20cd7b73a320d", algorithm=MD5, uri=" sip:15025482003@sip.gafachi.com <sip%3A15025482003@sip.gafachi.com>", response="617d7d2f882293a061ad27a72cd4ca60" Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 325 Remote-Party-ID: "" <sip:@sip.gafachi.com <sip%3A@sip.gafachi.com>
;screen=yes;privacy=off
v=0 o=FreeSWITCH 4716203835921160991 4794018502006774974 IN IP4 10.252.115.245 s=FreeSWITCH c=IN IP4 75.101.133.137 t=0 0 a=sendrecv m=audio 28950 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 On Thu, Jun 12, 2008 at 1:05 PM, Brian West <brian@freeswitch.org> wrote:
Show me the INVITE.
On Jun 12, 2008, at 11:50 AM, Nick Temple wrote:
SIP/2.0 604 Invalid Calling Party Number Via: SIP/2.0/UDP 75.101.133.137:5080 ;branch=z9hG4bKm4028av6cyD0F;received=75.101.133.137;rport=5080 From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com<sip%3Aa6031sTVQLeOhyIV@sip.gafachi.com> ;transport=udp>;tag=j8BBv5jaycaFF To: <sip:15025482003@sip.gafachi.com<sip%3A15025482003@sip.gafachi.com>
;tag=gss0c342e9c3ad2398e2f610f7ce613a089 Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Server: Gafachi UAS v110.08 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: <sip:15025482003@64.192.112.13<sip%3A15025482003@64.192.112.13>
Content-Length: 0
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
This is why. Seems your profile is missing the tech_pvt-
caller_profile->caller_id_number and tech_pvt->caller_profile- caller_id_name ..... Can you provide me an account and your js code so I can reproduce this?
/b On Jun 12, 2008, at 12:12 PM, Nick Temple wrote:
Remote-Party-ID: "" <sip:@sip.gafachi.com>;screen=yes;privacy=off
Brian West sip:brian@freeswitch.org
I really would like to see how you're doing this.. because if you're doing this from an javascript and its a single leg call you need to set origination_caller_id_name and origination_caller_id_number on the originate line inside the {}, On Jun 12, 2008, at 12:12 PM, Nick Temple wrote:
Remote-Party-ID: "" <sip:@sip.gafachi.com>;screen=yes;privacy=off
Brian West sip:brian@freeswitch.org
I know change the {} to [] this has to be a bug introduced in a recent patch in trunk. /b On Jun 12, 2008, at 12:12 PM, Nick Temple wrote:
Show me the INVITE:
INVITE sip:1502XXXXXXX@sip.gafachi.com SIP/2.0 Via: SIP/2.0/UDP 75.101.133.137:5080;rport;branch=z9hG4bKm4028av6cyD0F Max-Forwards: 70 From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com;transport=udp>;tag=j8BBv5jaycaFF To: <sip:15025482003@sip.gafachi.com> Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Contact: <sip:a6031sTVQLeOhyIV@75.101.133.137:5080>;transport=udp Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.0-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Allow-Events: talk Proxy-Authorization: Digest username="a6031sTVQLeOhyIV", realm="sip.gafachi.com", nonce="4fa20cd7b73a320d", algorithm=MD5, uri="sip:15025482003@sip.gafachi.com", response="617d7d2f882293a061ad27a72cd4ca60" Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 325 Remote-Party-ID: "" <sip:@sip.gafachi.com>;screen=yes;privacy=off
v=0 o=FreeSWITCH 4716203835921160991 4794018502006774974 IN IP4 10.252.115.245 s=FreeSWITCH c=IN IP4 75.101.133.137 t=0 0 a=sendrecv m=audio 28950 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
On Thu, Jun 12, 2008 at 1:05 PM, Brian West <brian@freeswitch.org> wrote: Show me the INVITE.
On Jun 12, 2008, at 11:50 AM, Nick Temple wrote:
SIP/2.0 604 Invalid Calling Party Number Via: SIP/2.0/UDP 75.101.133.137 :5080;branch=z9hG4bKm4028av6cyD0F;received=75.101.133.137;rport=5080 From: "" < sip:a6031sTVQLeOhyIV@sip.gafachi.com;transport=udp>;tag=j8BBv5jaycaFF To: <sip: 15025482003@sip.gafachi.com>;tag=gss0c342e9c3ad2398e2f610f7ce613a089 Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Server: Gafachi UAS v110.08 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: <sip:15025482003@64.192.112.13> Content-Length: 0
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
_______________________________________________ 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
Brian West sip:brian@freeswitch.org
This was the patch that did this I think. Can you revert that and try it? http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_ivr_originate.c?r... /b On Jun 12, 2008, at 12:12 PM, Nick Temple wrote:
Show me the INVITE:
INVITE sip:1502XXXXXXX@sip.gafachi.com SIP/2.0 Via: SIP/2.0/UDP 75.101.133.137:5080;rport;branch=z9hG4bKm4028av6cyD0F Max-Forwards: 70 From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com;transport=udp>;tag=j8BBv5jaycaFF To: <sip:15025482003@sip.gafachi.com> Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Contact: <sip:a6031sTVQLeOhyIV@75.101.133.137:5080>;transport=udp Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.0-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, timer Allow-Events: talk Proxy-Authorization: Digest username="a6031sTVQLeOhyIV", realm="sip.gafachi.com", nonce="4fa20cd7b73a320d", algorithm=MD5, uri="sip:15025482003@sip.gafachi.com", response="617d7d2f882293a061ad27a72cd4ca60" Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 325 Remote-Party-ID: "" <sip:@sip.gafachi.com>;screen=yes;privacy=off
v=0 o=FreeSWITCH 4716203835921160991 4794018502006774974 IN IP4 10.252.115.245 s=FreeSWITCH c=IN IP4 75.101.133.137 t=0 0 a=sendrecv m=audio 28950 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
On Thu, Jun 12, 2008 at 1:05 PM, Brian West <brian@freeswitch.org> wrote: Show me the INVITE.
On Jun 12, 2008, at 11:50 AM, Nick Temple wrote:
SIP/2.0 604 Invalid Calling Party Number Via: SIP/2.0/UDP 75.101.133.137 :5080;branch=z9hG4bKm4028av6cyD0F;received=75.101.133.137;rport=5080 From: "" < sip:a6031sTVQLeOhyIV@sip.gafachi.com;transport=udp>;tag=j8BBv5jaycaFF To: <sip: 15025482003@sip.gafachi.com>;tag=gss0c342e9c3ad2398e2f610f7ce613a089 Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 CSeq: 100525131 INVITE Server: Gafachi UAS v110.08 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Contact: <sip:15025482003@64.192.112.13> Content-Length: 0
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
_______________________________________________ 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
Brian West sip:brian@freeswitch.org
Nick: http://wiki.freeswitch.org/wiki/Special:Search?search=caller&go=Go You can do a "TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch", setting the environment variable for posting all SIP Messages to you that get passed or received by sofia. - http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP I would be interested in having Gafachi post an item in http://wiki.freeswitch.org/wiki/SIP_Provider_Examples Perhaps they can join us on IRC and we can help them put a listing on that page as soon as their interop works ? Matt On Wed, 11 Jun 2008, Nick Temple wrote:
Hello,
I am attempting to get Gafachi setup on our system. They are telling me that the caller id is not being sent with the request, and are thus rejecting the calls.
Is there a short FAQ on how to send the caller id?
in vars.xml, I have:
<X-PRE-PROCESS cmd="set" data="outbound_caller_name=My Name"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=XXXXXXXXXX"/>
call.js:
call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
sessionx = new Session(); rs = sessionx.originate(session, call, timeout);
I've also tried setting various forms of {caller_id_name}, {origination_caller_id_name}, {effective_caller_id_name} / number, but with little success.
I am unsure which of the above to send in this context.
Where in the trace / logs / session variables can I / should I see the caller_id being sent? Is there a setting I'm missing someplace?
I'm sure I've missed something simple.
thanks for your help,
Nick
I'll invite Gafachi to do so ... I've seen from the mailing list archives that other people *are* succesfully using Gafachi with Freeswitch, I'm assuming once I get this small issue resolved things will work. Thanks for the search pointer, I've been through the Wiki but this is bringing up additional pages I missed. Can you (or anyone) post someplace a trace of what a successful caller_id SIP message looks like? Maybe I'd be more successful if I new exactly what I was looking for? Thanks for the help, Nick On Thu, Jun 12, 2008 at 10:34 AM, Matt Klein <mklein@nmedia.net> wrote:
Nick:
http://wiki.freeswitch.org/wiki/Special:Search?search=caller&go=Go
You can do a "TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch", setting the environment variable for posting all SIP Messages to you that get passed or received by sofia. - http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP
I would be interested in having Gafachi post an item in http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
Perhaps they can join us on IRC and we can help them put a listing on that page as soon as their interop works ?
Matt
On Wed, 11 Jun 2008, Nick Temple wrote:
Hello,
I am attempting to get Gafachi setup on our system. They are telling me that the caller id is not being sent with the request, and are thus rejecting the calls.
Is there a short FAQ on how to send the caller id?
in vars.xml, I have:
<X-PRE-PROCESS cmd="set" data="outbound_caller_name=My Name"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=XXXXXXXXXX"/>
call.js:
call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/ sip.gafachi.com/1XXXXXXXXXX";
sessionx = new Session(); rs = sessionx.originate(session, call, timeout);
I've also tried setting various forms of {caller_id_name}, {origination_caller_id_name}, {effective_caller_id_name} / number, but with little success.
I am unsure which of the above to send in this context.
Where in the trace / logs / session variables can I / should I see the caller_id being sent? Is there a setting I'm missing someplace?
I'm sure I've missed something simple.
thanks for your help,
Nick
_______________________________________________ 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
participants (3)
-
Brian West
-
Matt Klein
-
Nick Temple