Re: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time
Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ 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
[Freeswitch-users] Force endpoint to use rfc2833 for dtmf
Hello, in a bigger installation with some thousand endpoints in the field we see, that the endpoint equipment is always using INFO messages (standard setting is auto, so the endpoint decides which method to use). I have 2 questions to that scenario: 1. Is there a way that Freeswitch forces/restricts the endpoint to use rfc2833 or not to send to allow INFO in the invite message? 2. Currently INFO messages do not get forwarded from the caller through freeswitch to called endpoint. How can we enable that FS is fowarding the INFO messages? Best regards Peter ___ 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
Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
I just crosschecked the dialplan which is used. We do not anwer the call, we bridge it directly to a PSTN destination. However the Ringing event is not passed to PSTN(A): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS PSTN(A)ACK==FS But then I stumbled over the following SOFIA LOOPBACK entry in the logs: 2009-12-21 12:47:00.404145 [DEBUG] switch_core_state_machine.c:351 (sofia/external/06322xxx...@10.11.12.15) State XCHANGE_MEDIA 2009-12-21 12:47:00.404145 [DEBUG] mod_sofia.c:469 SOFIA LOOPBACK 2009-12-21 12:47:00.404145 [DEBUG] sofia.c:3669 Channel sofia/external/0171...@10.11.12.15:5060 skipping state [early][183] So I modified the dialplan to temporarily use another Patton GW for outgoing calls, et voilà, I receive a ringing tone at PSTN(A). So I think this is because Freeswitch thinks this is a loopback, because incoming and outgoing gateway is the same. But I due to other restrictions we need the call to pass through the same Patton Gateway to PSTN(B) as we received it from PSTN(A). Is there a chance to tell Freeswitch to not consider this call as a loopback scenario? Best regards Peter Brian West schrieb: That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. /b On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: Should I open a JIRA for this? Best regards Peter ___ 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
Re: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header
we do this based XML-Curl. Jerry Richards schrieb: Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN
Should I open a JIRA for this? Best regards Peter Peter P GMX schrieb: Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS PSTN(A)ACK==FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter ___ 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
[Freeswitch-users] How to overcome 415 Unsupported Media Type
I try to attach Bravis video conference clients to Freeswitch. Those video conference clients are really working good (Multilingual clients for testing ca be downloaded here: http://www.bravis.eu/). Some big companies here in Germany use them in large installations. They are based on SIP, but do not use any publicly known codecs. Normally they are maintained and routed via our OpenSIPS server, but I would like to integrate them into our Freeswitch system. That way I do not have to manage 2 SIP servers for phone calls and video conferencing calls. However the SIP message does not provide Content-Type: application/sdp. Instead it provides Content-Type: application/BRAVIS. The clients register successfully but they do not invite. Freeswitch answers SIP/2.0 415 Unsupported Media Type. I have added bypass_media=true into the dialplan and inbound-late-negotiation true in the internal profile but this didn't help. I think Freeswitch complains about the content-type. Is there any way how I may overcome this? Here is a sample Invite INVITE sip:835...@sip5.mydomain.com SIP/2.0. From: myname sip:835...@sip5.mydomain.com;tag=5c5c3ef6bbe9de119f1aa11f7ca41a5f. To: sip:835...@sip5.mydomain.com. Via: SIP/2.0/UDP 217.xxx.xxx.xx6:5530;iid=9931;branch=z9hG4bKc4583ef6bbe9de119f1aa11f7ca41a5f;uas-addr=217.24.11.190;rport. CSeq: 4711 INVITE. Call-ID: 2-ee3d3ef6-bbe9-de11-9fa1-a11f7ca41a5f. Contact: myname sip:835...@217.xxx.xxx.xx6:5530. User-Agent: BRAVIS/1.5.20.27.4585 (Linux 2.6.31-16-generic; generic; Ubuntu 9.10; i686; de; 8). Max-Forwards: 70. Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS. Supported: 100rel. Content-Type: application/BRAVIS. Content-Length: 174. ACAABAAAFDAAABAAMEFBHCGLACAACAAACPKNBHGOAPLDABAAFAAADBABAAPPELAFAACAAAHDHCGGGMHIPPUPOPBEKHHHAPLDOPBEKHHHAPLDABAADCABAAADFBMDHOAEAAGIGPHDHEAAPPJFKGAPLHHNKF. Best regards Peter ___ 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
Re: [Freeswitch-users] Voicemail-Email
Hello Oliver,
I have the same on Ubuntu wth newest trunk.
Best regards
Peter
Oliver Schönbeck schrieb:
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get
the mod_voicemail to send the received messages to the user by using
exim4 on a debian machine.
So far we followed the instructions in the wiki article (
http://wiki.freeswitch.org/wiki/Mod_voicemail ).
I added some lines to the bash script to enable some kind of logging:
#! /bin/bash
typeset LOG=/tmp/${0##*/}.out
mv $LOG ${LOG}.old /dev/null 21
[[ -t 1 ]] echo Writing to logfile '$LOG'.
exec $LOG 21
exim4 -t -v $LOG
If I run the script from the command line everything is working as
expected. If the script gets called by freeswitch I get the following
result in my logfile:
/usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation
fault (core dumped) exim4 -t -v $LOG
Has anybody seen similar effects before?
Any advice whats going wrong is heavily appreciated.
Thanks
Oliver
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[Freeswitch-users] No Ringing tone when call is forwarded to PSTN
Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)INVITE===FS PSTN(A)===TRYING===FS FS===INVITE==PSTN(B) FS==TRYING===PSTN(B) FS==RINGING==PSTN(B) PSTN(A)==PROGRESS===FS FS===OK==PSTN(B) FSACKPSTN(B) PSTN(A)===OKFS PSTN(A)ACK==FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter ___ 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] Which ATAs to chose for modem connections?
We currently use Patton gateways SN4116 for attaching fax and modem equipment to our Freeswitch system. Freeswitch is in bypass-media-mode, so media flow goes the following way: Modem/Fax = Patton_SN4116 = Patton_SN46XX =PSTN/ISDN However modem connections are not very reliable. We exchanged the SN4118 against a Fritzbox ATA and the situation improves. However Fritzboxes do not deliver the number of ports we need. What is your experience? Which ATA is the best choice for modem connections? Best regards Peter ___ 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] Invite local number into a conference - codec problem
Hello, I try to invite a user into a conference by loopback/255 8000 Conference 255 is the user, I invite the user via loopback as that way I can also invite external numbers. It processes the user's local dialplan correctly (as if the user was normally dialled), however it only offers L16 codec, so the Phone fails. I can see no codec negociation on the debug console. If I call the phone from another phone, then codec negociation is taking place. If I invite an external PSTN user into the conference then codecs are set correctly (L16+PCMA+PCMU etc) Is there a way to explicitely set the codec for the conference? !--param name=disable-transcoding value=true/-- is not set is, still commented in the internal profile. In vars.conf.xml only only PCMA and PCMU are set. Best regards Peter ___ 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
Re: [Freeswitch-users] continue_on_fail
Hello Nandy, thanks for your hint, but it's a bit more than that. In our application which is handled via XML-Curl, the user can define it's forwards on a web interface. He can enter mixed local or external numbers which are called sequentially or in parallel. Best regards Peter Nandy Dagondon schrieb: this action can be accomplished using Group Dialing (Sequential). this may not answer your problem but have you considered it? -nandy On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: I have a Problem with continue_on_fail. I have setup a hunt group action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/ action application=bridge data=sofia/external/2...@10.11.12.243 mailto:2...@10.11.12.243,sofia/external/2...@10.11.12.234 mailto:2...@10.11.12.234,sofia/external/2...@10.11.12.188 mailto:2...@10.11.12.188,sofia/external/1...@10.11.12.245 mailto:1...@10.11.12.245/ action application=bridge data= (dialstring for fallback user ) I want the fallback user to be called whenever none of the previously called 3 gateway numbers picks up or if they are all busy. Therefore continue_on_fail=NO_ANSWER,USER_BUSY The fallback user is called, however if any of the previously called gateways picks up and then hangs up, the fallback user is called afterwards. Means: The fallback user is always called. I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire the next bridge if it gets a NORMAL_CLEARING. Am I thinking wrongly about this? I have added action application=set data=hangup_after_bridge=true/ and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 ___ 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] Multiple registration: Registration1 cannot call Registration2 in attended_transfer
Hello, in our dialplan we have enabled multiple-registrations, so 2 phones can register on a single directory entry. param name=multiple-registrations value=true/ Both phones are registered, both phones can be called and each phone can call the other phone. However in an attended_transfer mode calls cannot be transferred to the other phone with the same number. Attended_transfer in this case is needed when you take a call on your main SIP phone and and then want to transfer it to your mobile DECT/SIP phone, because you may have to check something in another room. I did a SIP trace and see the following: * A invites B(phone 1) = ok * B(phone 1) places call on hold = ok * B(phone 1) dials number B(phone 2 DECT) on second line * Freeswitch send Invite to B(phone 1) = ok * Freeswitch send Invite to B(phone 2 DECT) * B(phone 2 DECT) sends Ringing to Freeswitch = ok * B(phone 1) sends Busy to Freeswitch * B(phone 1) displays Busy and hangs up the second line Is there any way to overcome this? Is there a way to ignore the Busy from phone 1 when phone 2 answers Ringing? Best regards Peter ___ 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] Force presence status manually
Hello, is there a way to manually force a presence status update? In our scenario we have a Freeswitch cluster. As phones sometimes register on one and one time on another machine via the load balancer, we cannot dial via user/exten. Instead we dial each phone by it's register string via xml-curl. That way -when a phone is called - other phones who subscribed to this phone, do not receive a message to update their presence status. Is there a way to force the pesence status of a phone manually in the dialplan? We may then set the status before bridging and then reset it with a hangup hook. Best regards Peter ___ 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] continue_on_fail
I have a Problem with continue_on_fail. I have setup a hunt group action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/ action application=bridge data=sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245/ action application=bridge data= (dialstring for fallback user ) I want the fallback user to be called whenever none of the previously called 3 gateway numbers picks up or if they are all busy. Therefore continue_on_fail=NO_ANSWER,USER_BUSY The fallback user is called, however if any of the previously called gateways picks up and then hangs up, the fallback user is called afterwards. Means: The fallback user is always called. I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire the next bridge if it gets a NORMAL_CLEARING. Am I thinking wrongly about this? I have added action application=set data=hangup_after_bridge=true/ and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter ___ 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
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Hello,
i now changed the $${domain} name of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host, presence_hosts, server_user,server_host,
hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1;
+---+---+-+-++---+--+---+
| sip_host | presence_hosts| server_user | server_host |
hostname | sip_realm | mwi_user | mwi_host |
+---+---+-+-++---+--+---+
| sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 |
sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com |
+---+---+-+-++---+--+---+
IPs are:
10.11.12.1 sip1.mydomain.com (common cluster IP)
10.11.12.2 sip11.mydomain.com
10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is:
document type=freeswitch/xml
section name=directory
domain name=sip1.mydomain.com
user id=100
params
param name=password value=pass/
param name=vm-password value=pass/
param name=vm-email-all-messages value=true/
param name=vm-attach-file value=true/
param name=vm-mailto value=em...@domain.net/
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_us...@${dialed_domain})}/
param name=http-allowed-api value=voicemail/
/params
variables
variable name=accountcode value=800/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 100/
variable name=effective_caller_id_number value=100/
/variables
/user
/domain
/section
/document
The internal profile has the following alias:
profile name=internal domain=$${domain}
aliases
alias name=$${domain}/
alias name=sip1.mydomain.com/
alias name=default/
/aliases
With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI
does not. Any hint what I should change to make this work?
Best regards
Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com http://sip1.mydomain.com is the domain in the
packet and thus the profile should have an alias for this.
Then the user must reside in your sip db with the user 200 and domain
sip1.mydomain.com http://sip1.mydomain.com
if you dont have this consider the force-register-domain and
force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
Did you check the 2 replies that told you you need aliases in your
sofia profile to translate the domain found in your
message_waiting to the right profile? Both Brian and Mike
answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
I tried now with phones directly attached to the freeswitch
(without an
OpenSIPS in between). I also added the alias. But the
behaviour is as
before:
No notify message from freeswitch, neither after register nor
after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias will be required for every domain you run on
the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Hello,
i now changed the $${domain} of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works. Thanks to all for your support.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host, presence_hosts, server_user,server_host,
hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1;
+---+---+-+-++---+--+---+
| sip_host | presence_hosts| server_user | server_host |
hostname | sip_realm | mwi_user | mwi_host |
+---+---+-+-++---+--+---+
| sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 |
sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com |
+---+---+-+-++---+--+---+
IPs are:
10.11.12.1 sip1.mydomain.com (common cluster IP)
10.11.12.2 sip11.mydomain.com
10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is:
document type=freeswitch/xml
section name=directory
domain name=sip1.mydomain.com
user id=100
params
param name=password value=pass/
param name=vm-password value=pass/
param name=vm-email-all-messages value=true/
param name=vm-attach-file value=true/
param name=vm-mailto value=em...@domain.net/
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_us...@${dialed_domain})}/
param name=http-allowed-api value=voicemail/
/params
variables
variable name=accountcode value=800/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 100/
variable name=effective_caller_id_number value=100/
/variables
/user
/domain
/section
/document
The internal profile has the following alias:
profile name=internal domain=$${domain}
aliases
alias name=$${domain}/
alias name=sip1.mydomain.com/
alias name=default/
/aliases
With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI
does not. Any hint what I should change to make this work?
Best regards
Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com http://sip1.mydomain.com is the domain in the
packet and thus the profile should have an alias for this.
Then the user must reside in your sip db with the user 200 and domain
sip1.mydomain.com http://sip1.mydomain.com
if you dont have this consider the force-register-domain and
force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
Did you check the 2 replies that told you you need aliases in your
sofia profile to translate the domain found in your
message_waiting to the right profile? Both Brian and Mike
answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
I tried now with phones directly attached to the freeswitch
(without an
OpenSIPS in between). I also added the alias. But the
behaviour is as
before:
No notify message from freeswitch, neither after register nor
after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias will be required for every domain you run on
the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
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Re: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall???
Concerning, Which I'm kinda confused about, I don't have any 192.168 net here?? I think, this is a default entry in the acl.conf.xml. Please check the entries there. But normally this shouldn't stop freeswitch from working and handling requests. Can you set the console_log_level to debug in vars.xml and post you console output when the phone tries to register? You may also grep the network traffic on port 5060 (e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's wrong. Best regards Peter mailinglist schrieb: Hi Adam Excellent first steps! Thankyou for the hint. Now I hope somebody can tell me what I'm doing wrong next... I've gotten it to register to the testprovider here (musimi.dk), but I get an error when I create an account for testing with the X-Lite phone. It displays 403 forbidden in the display. I've created an account on FreeSwitch extension 1001 password 1001 mailbox 1001 voicemail password 1001 account code 1001 Effective Caller ID Name Fribert Effective Caller ID Number 4692 (the Musimi number) Voicemail Mail To my address Voicemail Attach File true User Context default Call Group Enabled true Extension Description Test number In the X-Lite Display Name Fribert User name 1001 Password 1001 Autorization user name 1001 Domain LAN-IP-OF-pfSense Check in Register with domain and receive incoming calls Check in domain. That's about it. Looking on the status page, I can see these lines in the log: 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding 192.168.42.0/24 (deny) to list lan 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding 192.168.42.42/32 (allow) to list lan 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 Which I'm kinda confused about, I don't have any 192.168 net here??? But as it also primarily forbids it, except .42 to allow, I'm wondering if it could be something internal? Best regards 05-12-2009 kl. 18:52 skrev Adam Ford li...@redbonez.net i meddelelsen 0e0013f55e224674a1361329cf7a8...@redbonez: I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I have found with a stable release. It was very easy to use, I would recommend it if you just want a quick base system with standard features. Though, I ended up switching to a compiled version of FreeSWITCH in order to make the customizations I needed for my office. http://doc.pfsense.org/index.php/FreeSWITCH -AF *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *mailinglist *Sent:* Saturday, December 05, 2009 2:47 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. ___ 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
[Freeswitch-users] uuid_broadcast with pause and rewind for dictation service?
Hello, I would like to offer a dictation service to a secretary. Means: * the boss is dictating some text on a certain phone number * the secretary picks up the recording on the phone and types the text into the computer As the secretary is not able to type in as fastly as heir boss is able to speak, she needs some kind of pause and rewind button. 1st question: Is there any functionality available for example in uuid_broadcast? 2nd question: How much would be the effort to implement this (uuid_broadcast_pause, uuid_broadcast_UNpause, uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. Best regards Peter ___ 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
Re: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service?
Oh that's a lot of money, anybody else needs this feature, so we may share a bounty? Best regards Peter Anthony Minessale schrieb: Someone else was asking about this too. I could probably write a dictaction mod in c like the one I made for asterisk starting at about $3k depending on the featureset required. On Dec 6, 2009 10:30 AM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: Hello, I would like to offer a dictation service to a secretary. Means: * the boss is dictating some text on a certain phone number * the secretary picks up the recording on the phone and types the text into the computer As the secretary is not able to type in as fastly as heir boss is able to speak, she needs some kind of pause and rewind button. 1st question: Is there any functionality available for example in uuid_broadcast? 2nd question: How much would be the effort to implement this (uuid_broadcast_pause, uuid_broadcast_UNpause, uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 ___ 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
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host, presence_hosts, server_user,server_host,
hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1;
+---+---+-+-++---+--+---+
| sip_host | presence_hosts| server_user | server_host |
hostname | sip_realm | mwi_user | mwi_host |
+---+---+-+-++---+--+---+
| sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 |
sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com |
+---+---+-+-++---+--+---+
IPs are:
10.11.12.1 sip1.mydomain.com (common cluster IP)
10.11.12.2 sip11.mydomain.com
10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is:
document type=freeswitch/xml
section name=directory
domain name=sip1.mydomain.com
user id=100
params
param name=password value=pass/
param name=vm-password value=pass/
param name=vm-email-all-messages value=true/
param name=vm-attach-file value=true/
param name=vm-mailto value=em...@domain.net/
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_us...@${dialed_domain})}/
param name=http-allowed-api value=voicemail/
/params
variables
variable name=accountcode value=800/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 100/
variable name=effective_caller_id_number value=100/
/variables
/user
/domain
/section
/document
The internal profile has the following alias:
profile name=internal domain=$${domain}
aliases
alias name=$${domain}/
alias name=sip1.mydomain.com/
alias name=default/
/aliases
With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI
does not. Any hint what I should change to make this work?
Best regards
Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com http://sip1.mydomain.com is the domain in the
packet and thus the profile should have an alias for this.
Then the user must reside in your sip db with the user 200 and domain
sip1.mydomain.com http://sip1.mydomain.com
if you dont have this consider the force-register-domain and
force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
Did you check the 2 replies that told you you need aliases in your
sofia profile to translate the domain found in your
message_waiting to the right profile? Both Brian and Mike
answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
I tried now with phones directly attached to the freeswitch
(without an
OpenSIPS in between). I also added the alias. But the
behaviour is as
before:
No notify message from freeswitch, neither after register nor
after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias will be required for every domain you run on
the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
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Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
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MSN:anthony_miness...@hotmail.com
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[Freeswitch-users] Voicmail - message only
Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ 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
Re: [Freeswitch-users] Voicmail - message only
I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to only play announcement or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ 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 ___ 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
Re: [Freeswitch-users] Voicmail - message only
Hello Anthony, thanks for the hint. I have posted a $100 bounty in the wiki + another $150 bounty to enable speaking an announcement via TTS. Best regards Peter Anthony Minessale schrieb: You could file it as a feature request and post a bounty and probably get the functionality fairly inexpensively maybe $100 On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: I would like to manage this in the voicemail menu. Press 6 to enable recording Press 7 to only play announcement or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 mailto: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 mailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded. Best regards Peter Brian West schrieb: Yes an alias will be required for every domain you run on the profile so it can find it. /b On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: Try an alias on the sip profile. Mike ___ 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
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set param name=manage-presence value=true/ in the internal profile. When a phone is set up I get the following Snom Phone REGISTER = OpenSIPS= Freeswitch Freeswitch OK = OpenSIPS=Snom Phone Snom Phone SUBSCRIBE = OpenSIPS= Freeswitch Freeswitch 202 Accepted = OpenSIPS=Snom Phone Snom Phone PUBLISH = OpenSIPS= Freeswitch Freeswitch 200 OK = OpenSIPS=Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswi...@sip11.mydomain.com nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:1...@sip1.mydomain.com SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_so...@192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_so...@192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7)
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Anthony, thanks for the hint, I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 2...@sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0) I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 2...@sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works. When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.cEvent-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent). Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI? Best regards Peter Anthony Minessale schrieb: connect to FS with fs_cli Issue the command: /events MESSAGE_QUERY MESSAGE_WAITING then leave some voice mails probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set param name=manage-presence value=true/ in the internal profile. When a phone is set up I get the following Snom Phone REGISTER = OpenSIPS= Freeswitch Freeswitch OK = OpenSIPS=Snom Phone Snom Phone SUBSCRIBE = OpenSIPS= Freeswitch Freeswitch 202 Accepted = OpenSIPS=Snom Phone Snom Phone PUBLISH = OpenSIPS= Freeswitch Freeswitch 200 OK = OpenSIPS=Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswi...@sip11.mydomain.com mailto:freeswi...@sip11.mydomain.com nta: received REGISTER sip:sip1.mydomain.com http://sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com http://sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params
Re: [Freeswitch-users] Problems with Voicemail
I sorted it out.
Something went wrong with the odbc database. I deleted the voicemail
database tables, restarted FS and let FS create the tables again. Now it
works.
I can even share the voicemails across 2 Freeswitch boxes.
Best regards
Peter
Peter P GMX schrieb:
I now created a file inbox.PCMA and get the following:
* inbox.PCMA is played
* the recorded voive mail file is not played (FS does not even try
to do that)
* then I hear
o to listen to the recording press 1
o to save the recording press 2
o ...
Here's the debug output
2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660
sofia/internal/2...@sip1.mydomain.com receive message [TRANSCODING_NECESSARY]
2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done
playing file
2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File
[/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done
playing file
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-listen_to_recording.wav] (en:en)
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec
Activated l...@8000hz 1 channels 20ms
2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660
sofia/internal/2...@sip1.mydomain.com receive message [TRANSCODING_NECESSARY]
2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done
playing file
nGrepping port 3306 I can see that the correct filenames are retrieved
from the mysql/odbc database:
1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor
Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.47
1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor
Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL.
Both filenames can be read.
Best regards
Peter
Peter P GMX schrieb:
I installed all sounds from SVN, but
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
isn't there. I checked another, older installation and couldn't this
file either.
I think that freeswitch tries to build a sound path for the file to be
played, and some parts of the path are missing.
I expect it would play a recorded message at that time in
/usr/local/freeswitch/storage/voicemail/default/${domain} and the
defined format is wav not pcma.
I also set storage_dir explicitely in the voicemail configs,but this
also didn't help.
Best regards
Peter
Brian West schrieb:
I'm going to venture to guess maybe the file was recorded in a
different codec and NOT pcma?
/b
On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote:
2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
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Re: [Freeswitch-users] Problems with Voicemail
I now created a file inbox.PCMA and get the following:
* inbox.PCMA is played
* the recorded voive mail file is not played (FS does not even try
to do that)
* then I hear
o to listen to the recording press 1
o to save the recording press 2
o ...
Here's the debug output
2009-11-22 20:17:43.701098 [DEBUG] switch_core_io.c:660
sofia/internal/2...@sip1.mydomain.com receive message [TRANSCODING_NECESSARY]
2009-11-22 20:17:44.278600 [DEBUG] switch_ivr_play_say.c:1428 done
playing file
2009-11-22 20:17:44.386776 [INFO] mod_native_file.c:82 Opening File
[/usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA] 8000hz
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1428 done
playing file
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:118 No language
specified - Using [en]
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:273 Handle
play-file:[voicemail/vm-listen_to_recording.wav] (en:en)
2009-11-22 20:17:45.201099 [DEBUG] switch_ivr_play_say.c:1136 Codec
Activated l...@8000hz 1 channels 20ms
2009-11-22 20:17:45.201099 [DEBUG] switch_core_io.c:660
sofia/internal/2...@sip1.mydomain.com receive message [TRANSCODING_NECESSARY]
2009-11-22 20:17:46.419933 [DEBUG] switch_ivr_play_say.c:1428 done
playing file
nGrepping port 3306 I can see that the correct filenames are retrieved
from the mysql/odbc database:
1258894746.0.200.sip1.mydomain.com$d11c2a74-d766-11de-997b-bd7aecdc2a16.Gor
Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_c57a5e84-d766-11de-997b-bd7aecdc2a16.wav.4..B_NORMAL.47
1258897120.0.200.sip1.mydomain.com$580dafee-d76c-11de-84d4-a1cd7fa320b3.Gor
Nico.061035013113.inboxq/usr/local/freeswitch/storage/voicemail/default/sip1.mydomain.com/200/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL.
Both filenames can be read.
Best regards
Peter
Peter P GMX schrieb:
I installed all sounds from SVN, but
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
isn't there. I checked another, older installation and couldn't this
file either.
I think that freeswitch tries to build a sound path for the file to be
played, and some parts of the path are missing.
I expect it would play a recorded message at that time in
/usr/local/freeswitch/storage/voicemail/default/${domain} and the
defined format is wav not pcma.
I also set storage_dir explicitely in the voicemail configs,but this
also didn't help.
Best regards
Peter
Brian West schrieb:
I'm going to venture to guess maybe the file was recorded in a
different codec and NOT pcma?
/b
On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote:
2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
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Re: [Freeswitch-users] Problems with Voicemail
I installed all sounds from SVN, but
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
isn't there. I checked another, older installation and couldn't this
file either.
I think that freeswitch tries to build a sound path for the file to be
played, and some parts of the path are missing.
I expect it would play a recorded message at that time in
/usr/local/freeswitch/storage/voicemail/default/${domain} and the
defined format is wav not pcma.
I also set storage_dir explicitely in the voicemail configs,but this
also didn't help.
Best regards
Peter
Brian West schrieb:
I'm going to venture to guess maybe the file was recorded in a
different codec and NOT pcma?
/b
On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote:
2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
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[Freeswitch-users] Problems with Voicemail
Hello, i have a couple of problems with voicemail. Voicemails are recorded but not played in any way. 1) when I call my voicemail, I can hear the number of new messages, but I canot not hear the recorded files itself. I hear the following * You have 1 urgent new message in forder inbox * You have 7 new messages in forder inbox * New message number 1 Jan 011970 at 1 am * (message is NOT played) * You have 1 urgent new message in forder inbox * You have 7 new messages in forder inbox * press 1 to listen, press 2 * (I press 1) * New message number 1 Jan 011970 at 1 am * (message is NOT played) * You have 1 urgent new message in forder inbox * You have 7 new messages in forder inbox * ... The voicemail files are stored in the file system as wav files and I can play them manually from the file system - so there is sound inside. 2) Another strange thing is that all recorded calls are announced with a date of 01.Jan.1970 although the databse shows correct values. 3) Alternatively playing it on the web Gui on http://fs.ip:8080/api/voicemail/web doesn't work either. Date is again 01.Jan.1970 and shown length of the file is always 00:00:00, although the database shows the correct number of seconds 4) Just to note that whenever I expect a recorded file to be played I see the following on the console 2009-11-21 00:17:02.50 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA In my installation Freeswitch is running in a cluster and voicemails are stored in a mysql database. read_epoch is always 0, so file seems that Freeswitch never reads and updates an entry. Grepping mysql however shows a number of queries against the database and also the filenames are correctly read (output of ngrep): select * from voicemail_msgs where username='200' and domain='sip11.mydomain.com' and read_epoch=0 order by read_flags, created_epoch 1258748304.0.200.sip11.mydomain.com$db2801c4-d611-11de-8c58-554df1d6d322.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_b0fbf9e6-d611-11de-8c58-554df1d6d322.wav.15..A_URGENT 1258746833.0.200.sip11.mydomain.com$6e486a2e-d60e-11de-bb97-eb22f15930a0.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_50e727c2-d60e-11de-bb97-eb22f15930a0.wav.7..B_NORMAL 1258748679.0.200.sip11.mydomain.com$bac4dd0c-d612-11de-9618-afbc82bc409a.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_9e7865c4-d612-11de-9618-afbc82bc409a.wav.13..B_NORMAL 1258749095.0.200.sip11.mydomain.com$b2376082-d613-11de-80e8-89d0ee29138d.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_a6caaefc-d613-11de-80e8-89d0ee29138d.wav.6..B_NORMAL 1258749417.0.200.sip11.mydomain.com$726b375c-d614-11de-bb4c-6d51cf20cc23.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_6777907a-d614-11de-bb4c-6d51cf20cc23.wav.5..B_NORMAL 1258750260.0.200.sip11.mydomain.com$68cecedc-d616-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_5b6afb6c-d616-11de-b8c8-69b0064d633e.wav.9..B_NORMAL 1258753767.0.200.sip11.mydomain.com$93657422-d61e-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip11.mydomain.com/200/msg_84c1c588-d61e-11de-b8c8-69b0064d633e.wav.10..B_NORMAL Here's the debug log: EXECUTE sofia/internal/2...@sip1.mydomain.com send_display(VM 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_dptools.c:703 sofia/internal/2...@sip1.mydomain.com receive message [DISPLAY] EXECUTE sofia/internal/2...@sip1.mydomain.com voicemail(check default sip11.mydomain.com 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_voicemail.c:799 [default] rwlock 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-hello.wav] (en:en) 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated l...@8000hz 1 channels 20ms 2009-11-20 23:16:36.392353 [DEBUG] switch_core_io.c:660 sofia/internal/2...@sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:37.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:37.712349 [DEBUG] switch_channel.c:182 sofia/internal/2...@sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.812353 [DEBUG] switch_channel.c:182 sofia/internal/2...@sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.942376 [DEBUG] switch_channel.c:182 sofia/internal/2...@sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:273
Re: [Freeswitch-users] att_xfer and Loopback
Hello Anthony,
I made a console trace today:
http://pastebin.freeswitch.org/11125
Different from the mail below, in this case A and C have voice.
Best regards
Peter
Anthony Minessale schrieb:
if you provide a console trace of both situations with console
loglevel debug and put them on pastebin i can tell you what's happening.
On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Thanks Anthony,
however this rather deteriorated the situation.
Now it works the following
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C both have no voice (and B is
dropped)
- When A hangs up, C hangs up
Before it did:
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B is
dropped
- When A hangs up, C hangs up
Best regards
Peter
Anthony Minessale schrieb:
hit send too soon
you want to set loopback_bowout=false
This keeps loopback from trying to destroy itself when it sees a
chance to cut out of the call path.
On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale
anthony.miness...@gmail.com
mailto:anthony.miness...@gmail.com
mailto:anthony.miness...@gmail.com
mailto:anthony.miness...@gmail.com wrote:
set/export the channel variable loopback_bowout=true so it's on
the loopback leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net
mailto:prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
- A calls B
- B enters *4 gets an announcement and enter digits for C (A
get MOH)
- C is called
- As soon as C picks up the call, A and C are connected
and B
is dropped
How it should work until here:
- A calls B
- B enters *4 gets an announcement and enter digits for C (A
get MOH)
- C is called
- As soon as C picks up the call, B and C are connected (A
still MOH)
The dial string for C is dynamic and dependent on certain
parameters,
therefore C must be called via Loopback in our scenario.
Here are the configs:
In dialplan for calling B:
anti-action application=bind_meta_app data=4 b b
execute_extension::attended_xfer XML default/
Dialplan for executing the att_xfer:
extension name=attended_xfer
condition field=destination_number
expression=^attended_xfer$
action application=set
data=continue_on_fail=true/
action application=read data=3 4
ivr/ivr-enter_ext.wav
attxfer_callthis 3 #/
action application=set
data=origination_cancel_key=#/
action application=att_xfer
data=loopback/${attxfer_callthis}/
/condition
/extension
So this is pretty standard, except the loopback. SVN is
15322.
Anybody has a solution for this?
Best regards
Peter
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Re: [Freeswitch-users] att_xfer and Loopback
Thanks Anthony,
however this rather deteriorated the situation.
Now it works the following
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C both have no voice (and B is
dropped)
- When A hangs up, C hangs up
Before it did:
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B is dropped
- When A hangs up, C hangs up
Best regards
Peter
Anthony Minessale schrieb:
hit send too soon
you want to set loopback_bowout=false
This keeps loopback from trying to destroy itself when it sees a
chance to cut out of the call path.
On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
set/export the channel variable loopback_bowout=true so it's on
the loopback leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
- A calls B
- B enters *4 gets an announcement and enter digits for C (A
get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B
is dropped
How it should work until here:
- A calls B
- B enters *4 gets an announcement and enter digits for C (A
get MOH)
- C is called
- As soon as C picks up the call, B and C are connected (A
still MOH)
The dial string for C is dynamic and dependent on certain
parameters,
therefore C must be called via Loopback in our scenario.
Here are the configs:
In dialplan for calling B:
anti-action application=bind_meta_app data=4 b b
execute_extension::attended_xfer XML default/
Dialplan for executing the att_xfer:
extension name=attended_xfer
condition field=destination_number
expression=^attended_xfer$
action application=set data=continue_on_fail=true/
action application=read data=3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 3 #/
action application=set data=origination_cancel_key=#/
action application=att_xfer
data=loopback/${attxfer_callthis}/
/condition
/extension
So this is pretty standard, except the loopback. SVN is 15322.
Anybody has a solution for this?
Best regards
Peter
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com
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IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
mailto:sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
http://iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
mailto:googletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com
mailto:msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
mailto:paypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
mailto:sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
http://iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
mailto:googletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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[Freeswitch-users] att_xfer and Loopback
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
- A calls B
- B enters *4 gets an announcement and enter digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B is dropped
How it should work until here:
- A calls B
- B enters *4 gets an announcement and enter digits for C (A get MOH)
- C is called
- As soon as C picks up the call, B and C are connected (A still MOH)
The dial string for C is dynamic and dependent on certain parameters,
therefore C must be called via Loopback in our scenario.
Here are the configs:
In dialplan for calling B:
anti-action application=bind_meta_app data=4 b b
execute_extension::attended_xfer XML default/
Dialplan for executing the att_xfer:
extension name=attended_xfer
condition field=destination_number expression=^attended_xfer$
action application=set data=continue_on_fail=true/
action application=read data=3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 3 #/
action application=set data=origination_cancel_key=#/
action application=att_xfer data=loopback/${attxfer_callthis}/
/condition
/extension
So this is pretty standard, except the loopback. SVN is 15322.
Anybody has a solution for this?
Best regards
Peter
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[Freeswitch-users] Force registering external gateways though OpenSIPS load balancer
Hello, in a freeswitch cluster (FS1 and FS2) behind an OpenSIPS I want Freeswitch to register to external gateways through the OpenSIPS load balancer, in order to later receive incoming calls through the load balancer. Is there a way to tell Freeswitch in it's Gateway definition to define an additional path (e.g. fs_path) for it's registration, so that all registrations go via the OpenSIPS load balancer? Current flow FS1(register) = external_gateway (1st FS Machine) external_gateway(invite) =FS1 FS2(register) = external_gateway (2nd FS Machine) external_gateway(invite) =FS2 Desired flow FS1(register) = OpenSIPS = external_gateway (1st FS Machine) external_gateway(invite) = OpenSIPS = FS1 FS2(register) = OpenSIPS = external_gateway (2nd FS Machine) external_gateway(invite) = OpenSIPS =FS2 I tried to add fs_path to the gateway definition, but had no success. It still registered directly. Maybe by adding via route information or modifying contact header? Best regards Peter. ___ 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
Re: [Freeswitch-users] T.38 via UPDATE request
Hello Michael this is a productive system, so I can currently not update to the current trunk. But the installed SVN is 14741. Were those changes after 14741? Then I'll try to find a timeslot at night in order to update freeswitch. But Freeswitch should forward the UPDATE request in proxy-media mode, right? Best regards Peter Michael Jerris schrieb: There was just a bunch of work on UPDATE, can you confirm this is the same behavior with trunk? On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote: Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter ___ 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
[Freeswitch-users] T.38 via UPDATE request
Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter Here's the UPDATE message: UPDATE sip:mod_so...@82.115.xx.xxx:5080 SIP/2.0. Call-ID: 5283fe4e-334f-122d-d1b9-001517956764. Contact: sip:212.91.xxx.xxx:5060. Content-Type: application/sdp. CSeq: 16340063 UPDATE. From: sip:06912345...@sip.telco.de;tag=00-08135-017e041a-4d21f6037. Max-Forwards: 31. Route: sip:82.115.96.165;lr;ftag=pH663F4S02erm. To: 030987654321 sip:030987654...@82.115.xx.xxx;tag=pH663F4S02erm. User-Agent: Cirpack/v4.41e (gw_sip). Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31. Content-Length: 300. . v=0. o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx. s=SIP Call. c=IN IP4 212.91.xxx.xx. t=0 0. m=image 6860 udptl t38. a=sendrecv. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1000. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. ___ 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
Re: [Freeswitch-users] Mod_fifo posision in queue
Has anybody managed to get this to work already? How do you play the announcements dependent on the variable in the dialplan? Best regards Peter Michael Collins schrieb: On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola diego.vi...@gmail.com mailto:diego.vi...@gmail.com wrote: Lets make sure we add it on the wiki too =D. Yep, as soon as we verify its functionality we'll wikify it. :) -MC ___ 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
[Freeswitch-users] xml_curl configuration for failover cluster
Hello, I read in the wiki that binding blocks are processed in sequential order in a failover matter. So I created the following bindings for XML-Curl: However grepping the network traffic I can see that Freewitch always fetches both servers fo one binding. So there is no real failover. How can I avoid that? Best regards Peter configuration name=xml_curl.conf description=CURL XML Gateway bindings !-- Configuration -- !-- FIRST Application server -- binding name=configuration param name=gateway-url value=http://localhost/xml_curls/configuration; bindings=configuration/ /binding !-- SECOND Application server -- binding name=configuration_backup param name=gateway-url value=http://10.0.0.104/xml_curls/configuration; bindings=configuration/ /binding !-- Directory -- !-- FIRST Application server -- binding name=directory param name=gateway-url value=http://localhost/xml_curls/directory; bindings=directory/ /binding !-- SECOND Application server -- binding name=directory_backup param name=gateway-url value=http://10.0.0.104/xml_curls/directory; bindings=directory/ /binding !-- Dialplan -- !-- FIRST Application server -- binding name=dialplan param name=gateway-url value=http://localhost/xml_curls/dialplan; bindings=dialplan/ /binding !-- SECOND Application server -- binding name=dialpla_backup param name=gateway-url value=http://10.0.0.104/xml_curls/dialplan; bindings=dialplan/ /binding /bindings /configuration ___ 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] Siptapi and Freeswitch
Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter ___ 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
Re: [Freeswitch-users] Hangup: Always the same Q.850 cause code
: The Guy In IRC IS WRONG
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Reason: Q.850;cause=19;text=NO_ANSWER
Content-Length: 0
extension name=7016
condition field=destination_number expression=^7016$
action application=set data=sip_ignore_remote_cause=true/
action application=hangup data=user_busy/
/condition
/extension
send 630 bytes to udp/[72.128.89.126]:42988 at 14:35:31.286436:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP
10.0.1.8:50606;branch=z9hG4bK-d8754z-223ae00e1829097e-1---d8754z-;rport=42988;received=72.128.89.126
From: tonysip:t...@deathstar.freeswitch.org
mailto:sip%3at...@deathstar.freeswitch.org;tag=aa3b2b1d
To: 7016 sip:7...@deathstar.freeswitch.org
mailto:sip%3a7...@deathstar.freeswitch.org;tag=j4Q71UcUvvmcK
Call-ID: NDcyNmQyYjY5YWQwOTI3MjZiZWFlZDQyNDIyZjZlMDA.
CSeq: 1 INVITE
User-Agent: The Guy In IRC IS WRONG
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Reason: Q.850;cause=17;text=USER_BUSY
Content-Length: 0
On Fri, Sep 18, 2009 at 2:53 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Hello ,
I try to hangup aa call with a certain cause code.
If the user dials a number which we currently do not serve we send
action application=set data=sip_ignore_remote_cause=true/
action application=hangup data=NO_ANSWER/
which gives a
SIP/2.0 480 Temporarily Unavailable. Message , which is fine.
For the target number being busy or having another state, we use this.
anti-action application=set
data=sip_ignore_remote_cause=true/
anti-action application=hangup data=${hangup_cause}/
which gives a
SIP/2.0 486 Busy Here. Message , which is fine in case of Busy.
However in both cases the SIP mssage has the following cause code:
Reason: Q.850;cause=16;text=NORMAL_CLEARING.
which can lead to problems when forwarding to a PSTN Gateway.
How can we achieve, that the cause code is in sync with the deiivered
message?
We are on Trunk 14741M.
Best regards
Peter
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Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com
mailto:msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
mailto:paypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
mailto:sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
http://iax:gu...@conference.freeswitch.org/888
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pstn:213-799-1400
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Re: [Freeswitch-users] Hangup: Always the same Q.850 cause code
Hello,
I finally solved it by using
action application=hangup data=${originate_disposition}/
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did further testing on a second machine and found out the following:
After
action application=set data=sip_ignore_remote_cause=true/
action application=hangup data=NO_ANSWER/
The called party receives a NO_ANSWER
and the calling party receives a NORMAL_CLEARING
See the logs:
Best regards
Peter
Logs:
To called party:
U 82.xxx.9xx.163:5080 - 82.xxx.9xx.165:5060
CANCEL sip:0x...@21x.xx.xx.189:3273;line=fihb87zs SIP/2.0.
Via: SIP/2.0/UDP 82.xxx.9xx.163:5080;rport;branch=z9hG4bKg5SZ7829tHDae.
Route: sip:82.xxx.9xx.165.
Max-Forwards: 68.
From: 0x298 sip:0x...@82.xxx.9xx.162;tag=1mFgvS7t9Krtj.
To: sip:0x...@21x.xx.xx.189:3273;line=fihb87zs.
Call-ID: 9aab911f-22ce-122d-8686-001517956764.
CSeq: 120732503 CANCEL.
Reason: Q.850;cause=19;text=NO_ANSWER.
Content-Length: 0.
To calling party:
U 82.xxx.9xx.163:5062 - 82.xxx.9xx.165:5060
SIP/2.0 480 Temporarily Unavailable.
Via: SIP/2.0/UDP 82.xxx.9xx.165;branch=z9hG4bKc08d.b2a0b296.0.
Via: SIP/2.0/UDP
21x.xx.xx.189:2048;received=21x.xx.xx.189;branch=z9hG4bK-dnhr44fkakhd;rport=2048.
From: 0x298 sip:0x...@mydomain.de;tag=nvxy9h3rsk.
To: sip:0x...@mydomain.de;user=phone;tag=5y6B9FS9ZeUZB.
Call-ID: 3c49ea1f4563-8c3hia75cxuh.
CSeq: 2 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14741M.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Reason: Q.850;cause=16;text=NORMAL_CLEARING.
Content-Length: 0.
2009-09-23 12:51:37.008546 [NOTICE] switch_ivr_originate.c:2025 Hangup
sofia/external/0x...@21x.xx.xx.189:3273 [CS_CONSUME_MEDIA]
[NO_ANSWER]
2009-09-23 12:51:37.008546 [DEBUG] switch_channel.c:1715 Send signal
sofia/external/0x...@21x.xx.xx.189:3273 [KILL]
2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/0x...@21x.xx.xx.189:3273 [BREAK]
2009-09-23 12:51:37.008546 [DEBUG] switch_ivr_originate.c:2169 Originate
Resulted in Error Cause: 19 [NO_ANSWER]
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398
(sofia/external/0x...@21x.xx.xx.189:3273) Running State Change
CS_HANGUP
2009-09-23 12:51:37.008546 [INFO] mod_dptools.c:2098 Originate Failed.
Cause: NO_ANSWER
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434
(sofia/external/0x...@21x.xx.xx.189:3273) State HANGUP
EXECUTE sofia/internal/0x...@mydomain.de
set(sip_ignore_remote_cause=true)
2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:338 Channel
sofia/external/0x...@21x.xx.xx.189:3273 hanging up, cause: NO_ANSWER
2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:386 Sending CANCEL to
sofia/external/0x...@21x.xx.xx.189:3273
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:46
sofia/external/0x...@21x.xx.xx.189:3273 Standard HANGUP, cause:
NO_ANSWER
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434
(sofia/external/0x...@21x.xx.xx.189:3273) State HANGUP going to
sleep
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:479
(sofia/external/0x...@21x.xx.xx.189:3273) State Change CS_HANGUP
- CS_REPORTING
2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/0x...@21x.xx.xx.189:3273 [BREAK]
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398
(sofia/external/0x...@21x.xx.xx.189:3273) Running State Change
CS_REPORTING
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616
(sofia/external/0x...@21x.xx.xx.189:3273) State REPORTING
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:53
sofia/external/0x...@21x.xx.xx.189:3273 Standard REPORTING,
cause: NO_ANSWER
2009-09-23 12:51:37.008546 [DEBUG] mod_dptools.c:748
sofia/internal/0x...@mydomain.de SET
[sip_ignore_remote_cause]=[true]
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616
(sofia/external/0x...@21x.xx.xx.189:3273) State REPORTING going
to sleep
2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:411
(sofia/external/0x...@21x.xx.xx.189:3273) State Change
CS_REPORTING - CS_DESTROY
2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/0x...@21x.xx.xx.189:3273 [BREAK]
2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:1068 Session
129 (sofia/external/0x...@21x.xx.xx.189:3273) Locked, Waiting on
external entities
2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1086 Session
129 (sofia/external/0x...@21x.xx.xx.189:3273) Ended
2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1088 Close
[Freeswitch-users] Hangup: Always the same Q.850 cause code
Hello ,
I try to hangup aa call with a certain cause code.
If the user dials a number which we currently do not serve we send
action application=set data=sip_ignore_remote_cause=true/
action application=hangup data=NO_ANSWER/
which gives a
SIP/2.0 480 Temporarily Unavailable. Message , which is fine.
For the target number being busy or having another state, we use this.
anti-action application=set data=sip_ignore_remote_cause=true/
anti-action application=hangup data=${hangup_cause}/
which gives a
SIP/2.0 486 Busy Here. Message , which is fine in case of Busy.
However in both cases the SIP mssage has the following cause code:
Reason: Q.850;cause=16;text=NORMAL_CLEARING.
which can lead to problems when forwarding to a PSTN Gateway.
How can we achieve, that the cause code is in sync with the deiivered
message?
We are on Trunk 14741M.
Best regards
Peter
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Re: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes]
Thanks Anthony, that did the trick. Best regards Peter Anthony Minessale schrieb: you can edit mod_xml_curl.c line 64 and increase XML_CURL_MAX_BYTES On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: Hello, in a B2BUA scenario we have 2000 defined gateways (defined but not registered yet). When reloading mod_sofia Freeswitch complains about the XML-Curl File size 1MB and deactivates all gateways: mod_xml_curl.c:121 Oversized file detected [1056100 bytes] Is there any way to overcome this? Currently we have 2000 gateways defined. Finally we will have about 10.000. And we will not be able to reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. Best Regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
[Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes]
Hello, in a B2BUA scenario we have 2000 defined gateways (defined but not registered yet). When reloading mod_sofia Freeswitch complains about the XML-Curl File size 1MB and deactivates all gateways: mod_xml_curl.c:121 Oversized file detected [1056100 bytes] Is there any way to overcome this? Currently we have 2000 gateways defined. Finally we will have about 10.000. And we will not be able to reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. Best Regards Peter ___ 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
Re: [Freeswitch-users] Error building FreeSWITCH
I had the same problem. Must have been changed something in lua since this morning. Please install swig. E.g. on Debian sudo apt-get install swig That did it for me. Best regards Peter Lars Zeb schrieb: I just updated using “svn up” which brought the source to 14741. After running “./configure”, I ran “make” and got the following output: making all mod_lua make[5]: swig: Command not found make[5]: *** [mod_lua_wrap.cpp] Error 127 make[4]: *** [all] Error 1 make[3]: *** [mod_lua-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 What did I do wrong? Thanks, Lars ___ 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
Re: [Freeswitch-users] sofia_reg_external in odbc?
Hello Brian, I've done this. FS creates the tables sccessfully, but then doesn't fill them. isql: SQL show tables; +-+ | Tables_in_fs_external | +-+ | sip_authentication | | sip_dialogs | | sip_presence | | sip_registrations | | sip_shared_appearance_dialogs | | sip_shared_appearance_subscriptions | | sip_subscriptions | +-+ SQLRowCount returns 7 7 rows fetched Is that right, that the tables have the same structure as for the internal database? sofia status shows 7 registered external gateways, but none of them is shown in the ODBC database. All tables are empty. Any idea? Best regrads Peter Brian West schrieb: param name=odbc-dsn value=dsn:user:pass/ On the profile. /b On Aug 30, 2009, at 5:25 PM, Peter P GMX wrote: Hello, is there a chance to have sofia_reg_external in odbc/mysql instead of sqlite? In a B2BUA environment we have thousand of external registrations during a migration phase, and it would be good to have easy external control over the registered gateways (like in fs_internal. sip_registrations). Best regards Peter ___ 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
Re: [Freeswitch-users] SRTP Encryption
Sure this works, you can set rtp or srtp independently to every call leg (if FS is in media path) and even mix them in a conference. Best regards Peter NOx-WHV schrieb: Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. action application=set data=sip_secure_media=false/ It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone --(SRTP)-- Freeswitch --(RTP)-- Gateway Thanks for help NOx ___ 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
Re: [Freeswitch-users] SRTP Encryption
If you do not explicitely set bypass_media to true, then FS is in the media path. Best regards Peter NOx-WHV schrieb: How can I see if the FS is in media path? Or how can i set the FS in media path? Peter P GMX wrote: Sure this works, you can set rtp or srtp independently to every call leg (if FS is in media path) and even mix them in a conference. Best regards Peter NOx-WHV schrieb: Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. action application=set data=sip_secure_media=false/ It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone --(SRTP)-- Freeswitch --(RTP)-- Gateway Thanks for help NOx ___ 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
[Freeswitch-users] sofia_reg_external in odbc?
Hello, is there a chance to have sofia_reg_external in odbc/mysql instead of sqlite? In a B2BUA environment we have thousand of external registrations during a migration phase, and it would be good to have easy external control over the registered gateways (like in fs_internal. sip_registrations). Best regards Peter ___ 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
Re: [Freeswitch-users] sofia profile external register gwname via XML-Curl?
I got it, gateways have to be preloaded (rescanned) before they can be registered. Best regards peter Peter P GMX schrieb: Hello, I am using XML-Curl to handle the configuration of freeswitch When I try to register a gateway via event-socket with sofia profile external register gw-name I receive back invalid gateway. After reload mod_sofia the gateway is there. Question: Does this command work with xml-curl or only with local files?? At least I see no xml-curl request when grepping network traffic. Best regards Peter ___ 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
[Freeswitch-users] Calls from registered gateway try to lookup Directory
I have found a strange thing in my FS installation, FS is registered via a Gateway to an external provider (QSC) in the external context. But when a call is coming in, FS does not seem to go to any context, but tries to lookup the user, as I receive the following message 2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find user [026xx...@my.domain@my.domain] You must define a domain called 'my.domain' in your directory and add a user with the id=026xx...@my.domain attribute and you must configure your device to use the proper domain in it's authentication credentials. I learnt that a call from an external gateway should go to the public context. But (in CLI debug mode) there are no other messages, except the 3 lines above. What am I doing wrong? Best regards Peter Here the invite message. INVITE sip:gw+gw_xxx...@xx.xxx.xx.xxx:5080;transport=udp SIP/2.0. Via:SIP/2.0/UDP 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-. From:0XXsip:0xxx...@62.206.3.xxx;user=phone;tag=1616003581-1251392025611-. To:Mesip:026xx...@my.domain. Call-ID:bw185345611270809356816...@62.206.3.xxx. CSeq:778271239 INVITE. Contact:sip:62.206.3.xxx:5060. Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE. Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp. Supported:. Max-Forwards:20. Proxy-Authorization:DIGEST cnonce=fyvqi2pf,qop=auth,uri=sip:gw+gw_026xx...@xx.xxx.xx.xxx:5080;transport=udp,realm=my.domain,username=026xx...@my.domain,nonce=21bbe70c-932a-11de-b94d-bbade892ded3,algorithm=MD5,response=6d39a2546a4aa9a1fc39e2dc07c1e934,nc=0001. Content-Type:application/sdp. Content-Length:344. . v=0. o=BroadWorks 1271473 1 IN IP4 87.234.9.178. s=-. c=IN IP4 87.234.9.178. t=0 0. m=audio 18534 RTP/AVP 8 0 2 99 18 110. a=rtpmap:99 G726-24/8000. a=rtpmap:110 X-NSE/8000. a=fmtp:110 192-194,200-202. a=X-sqn:0. a=X-cap: 1 audio RTP/AVP 110. a=X-cpar: a=rtpmap:110 X-NSE/8000. a=X-cpar: a=fmtp:110 192-194,200-202. a=X-cap: 2 image udptl t38. ___ 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
Re: [Freeswitch-users] Calls from registered gateway try to lookup Directory
And yes, external profile is on Port 5080 and all request go to 5080. Best regards Peter Peter P GMX schrieb: I have found a strange thing in my FS installation, FS is registered via a Gateway to an external provider (QSC) in the external context. But when a call is coming in, FS does not seem to go to any context, but tries to lookup the user, as I receive the following message 2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find user [026xx...@my.domain@my.domain] You must define a domain called 'my.domain' in your directory and add a user with the id=026xx...@my.domain attribute and you must configure your device to use the proper domain in it's authentication credentials. I learnt that a call from an external gateway should go to the public context. But (in CLI debug mode) there are no other messages, except the 3 lines above. What am I doing wrong? Best regards Peter Here the invite message. INVITE sip:gw+gw_xxx...@xx.xxx.xx.xxx:5080;transport=udp SIP/2.0. Via:SIP/2.0/UDP 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-. From:0XXsip:0xxx...@62.206.3.xxx;user=phone;tag=1616003581-1251392025611-. To:Mesip:026xx...@my.domain. Call-ID:bw185345611270809356816...@62.206.3.xxx. CSeq:778271239 INVITE. Contact:sip:62.206.3.xxx:5060. Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE. Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp. Supported:. Max-Forwards:20. Proxy-Authorization:DIGEST cnonce=fyvqi2pf,qop=auth,uri=sip:gw+gw_026xx...@xx.xxx.xx.xxx:5080;transport=udp,realm=my.domain,username=026xx...@my.domain,nonce=21bbe70c-932a-11de-b94d-bbade892ded3,algorithm=MD5,response=6d39a2546a4aa9a1fc39e2dc07c1e934,nc=0001. Content-Type:application/sdp. Content-Length:344. . v=0. o=BroadWorks 1271473 1 IN IP4 87.234.9.178. s=-. c=IN IP4 87.234.9.178. t=0 0. m=audio 18534 RTP/AVP 8 0 2 99 18 110. a=rtpmap:99 G726-24/8000. a=rtpmap:110 X-NSE/8000. a=fmtp:110 192-194,200-202. a=X-sqn:0. a=X-cap: 1 audio RTP/AVP 110. a=X-cpar: a=rtpmap:110 X-NSE/8000. a=X-cpar: a=fmtp:110 192-194,200-202. a=X-cap: 2 image udptl t38. ___ 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] sofia profile external register gwname via XML-Curl?
Hello, I am using XML-Curl to handle the configuration of freeswitch When I try to register a gateway via event-socket with sofia profile external register gw-name I receive back invalid gateway. After reload mod_sofia the gateway is there. Question: Does this command work with xml-curl or only with local files?? At least I see no xml-curl request when grepping network traffic. Best regards Peter ___ 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] XML-RPC on different ip than 0.0.0.0
Hello, is there any chance to limit the listening ips of the xml-rpc server (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)? Best regards Peter ___ 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
Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hello Anthony, I set p...@30i,p...@30i and I can see in the logs that PCMA is used. However ptime is set to 20 msec as shown in the Logs: 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP: v=0 o=user 2075230 2075230 IN IP4 217.xx.xx.xxx s=call c=IN IP4 217.xx.xx.xxx t=0 0 m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101 a=rtpmap:2 G726-32/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/02xx...@fs1.my.domain) State NEW 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[G722:9:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec sofia/internal/02xx...@fs1.my.domain PCMA/8000 20 ms 160 samples 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf payload to 101 Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible. Best regards Peter Anthony Minessale schrieb: try setting FS to 30ms too edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks like p...@30i from: X-PRE-PROCESS cmd=set data=global_codec_prefs=g7...@32000h,g7...@16000h,G722,PCMU,PCMA,GSM/ X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMU,PCMA,GSM/ to: X-PRE-PROCESS cmd=set data=global_codec_prefs=g7...@32000h,g7...@16000h,G722,p...@30i,p...@30i,GSM/ X-PRE-PROCESS cmd=set data=outbound_codec_prefs=p...@30i,p...@30i,GSM/ On Fri, Aug 21, 2009 at 1:38 PM, Brian West br...@freeswitch.org mailto:br...@freeswitch.org wrote: You can ship me one whois bkw.org http://bkw.org, I can add it to my lab. /b On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote: BTW: We can ship you a FritzBox if you need one for testing. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hello Michael, I made some tests with Freeswitch and Fritzbox and found by Wireshark that: within one call * Freeswitch starts sending 20msec packets, then after ~0,2 second sends 30msec packets * FritzBox always sends 30msec packets. The average jitter is below 2 msec in both directions. The below logs shows that Freeswitch considers the FritzBox to be broken and starts using 30msec packets. But there is no SIP message from FS to Fritzbox telling him that FB will use 30msec packets. SDP from FS to Fritzbox always shows ptime:20 BTW: We can ship you a FritzBox if you need one for testing. Best regards Peter Log: 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP: v=0 o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx s=FreeSWITCH c=IN IP4 182.xxx.xx.xxx t=0 0 m=audio 30290 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer sofia/internal/02xx...@fs1.my.domain! 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/02xx...@fs1.my.domain [BREAK] EXECUTE sofia/internal/02xx...@fs1.my.domain playback(voicemail/8000/vm-that_was_an_invalid_ext.wav) 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated l...@8000hz 1 channels 20ms 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel sofia/internal/02xx...@fs1.my.domain entering state [early][183] 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649 sofia/internal/02xx...@fs1.my.domain receive message [TRANSCODING_NECESSARY] 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. ___ 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
Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hello Mathieu,
thank for your help. But this however didn't change the behaviour.
I've read of a patch in mod_sofia.c which partly corrects the problem
temporarily:
When I change Line 784 to
if (switch_rtp_ready(tech_pvt-rtp_session) codec_ms != tech_pvt-codec_ms) {
to
if (switch_rtp_ready(tech_pvt-rtp_session) codec_ms != tech_pvt-codec_ms
0) {
(add a 0) to deactivate this expression)
the announcements are played correctly to the Fritzbox. Connections to
other SIP phones (Snom) are also fine.
However the person at the Fritzbox still sounds very choppy in a
conference, but this is another module where I do not have a patch
available.
Best regards
Peter
Mathieu Rene schrieb:
Try setting that in your sip profile:
param name=rtp-autofix-timing value=false /
Thats a feature to work around with devices lying about their ptime in
their sdp payload.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
Hello Michael,
I made some tests with Freeswitch and Fritzbox and found by
Wireshark that:
within one call
* Freeswitch starts sending 20msec packets, then after ~0,2 second
sends 30msec packets
* FritzBox always sends 30msec packets.
The average jitter is below 2 msec in both directions.
The below logs shows that Freeswitch considers the FritzBox to be
broken
and starts using 30msec packets. But there is no SIP message from FS
to
Fritzbox telling him that FB will use 30msec packets. SDP from FS to
Fritzbox always shows ptime:20
BTW: We can ship you a FritzBox if you need one for testing.
Best regards
Peter
Log:
2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
v=0
o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
s=FreeSWITCH
c=IN IP4 182.xxx.xx.xxx
t=0 0
m=audio 30290 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
sofia/internal/02xx...@fs1.my.domain!
2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send
signal
sofia/internal/02xx...@fs1.my.domain [BREAK]
EXECUTE sofia/internal/02xx...@fs1.my.domain
playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
Activated l...@8000hz 1 channels 20ms
2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
sofia/internal/02xx...@fs1.my.domain entering state [early][183]
2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
sofia/internal/02xx...@fs1.my.domain receive message
[TRANSCODING_NECESSARY]
2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to
use
ptime 20 but what they meant to say was 30
This issue has so far been identified to happen on the following
broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so
broken
who knows what will happen..
___
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[Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hello, when calling from Fritzbox to a Snom Phone , sound is fine. But when calling an internal Freeswitch number (conference, mailbox) i hear a very choppy voice coming from the fritzbox side. I think it may have to do with the ptime 20msec/30msec. Example: When calling from the fritzbox to a voicemail then the annoucement from Freeswitch is choppy (too slow with interrups), but the recorded message is fine. Did anybody experience the same problem? Best regards Peter Here are some SIP messages: U 112.xxx.xx.xxx:5060 - 182.xxx.xx.xxx:5060 INVITE sip:0123...@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To: sip:0123...@my.domain. Call-ID: 3125316c8a2a1...@112.xxx.xx.xxx. CSeq: 55 INVITE. Contact: sip:02xx...@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3. Max-Forwards: 70. Expires: 120. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Supported: 100rel,replaces,timer. Allow-Events: telephone-event,refer. Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH. Content-Type: application/sdp. Accept: application/sdp, multipart/mixed. Accept-Encoding: identity. Content-Length: 359. . v=0. o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx. s=call. c=IN IP4 112.xxx.xx.xxx. t=0 0. m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101. a=sendrecv. a=rtpmap:2 G726-32/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=rtcp:7079. # U 182.xxx.xx.xxx:5060 - 112.xxx.xx.xxx:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To: sip:0123...@my.domain. Call-ID: 3125316c8a2a1...@112.xxx.xx.xxx. CSeq: 55 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Content-Length: 0. . # U 182.xxx.xx.xxx:5060 - 112.xxx.xx.xxx:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To: sip:0123...@my.domain;tag=7t1e8BQg5B7yK. Call-ID: 3125316c8a2a1...@112.xxx.xx.xxx. CSeq: 55 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Proxy-Authenticate: Digest realm=my.domain, nonce=900b46a0-8d88-11de-a6a1-098738f35adb, algorithm=MD5, qop=auth. Content-Length: 0. . # U 112.xxx.xx.xxx:5060 - 182.xxx.xx.xxx:5060 ACK sip:0123...@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To: sip:0123...@my.domain;tag=7t1e8BQg5B7yK. Call-ID: 3125316c8a2a1...@112.xxx.xx.xxx. CSeq: 55 ACK. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Content-Length: 0. . # U 112.xxx.xx.xxx:5060 - 182.xxx.xx.xxx:5060 INVITE sip:0123...@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To: sip:0123...@my.domain. Call-ID: 3125316c8a2a1...@112.xxx.xx.xxx. CSeq: 56 INVITE. Contact: sip:02xx...@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3. Proxy-Authorization: Digest username=02x, realm=my.domain, nonce=900b46a0-8d88-11de-a6a1-098738f35adb, uri=sip:0123...@my.domain, response=276b44e261c13bd17218adff1150f414, algorithm=MD5, cnonce=CADBE5D624516E8A, qop=auth, nc=0001. Max-Forwards: 70. Expires: 120. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Supported: 100rel,replaces,timer. Allow-Events: telephone-event,refer. Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH. Content-Type: application/sdp. Accept: application/sdp, multipart/mixed. Accept-Encoding: identity. Content-Length: 359. . v=0. o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx. s=call. c=IN IP4 112.xxx.xx.xxx. t=0 0. m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101. a=sendrecv. a=rtpmap:2 G726-32/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=rtcp:7079. # U 182.xxx.xx.xxx:5060 - 112.xxx.xx.xxx:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To: sip:0123...@my.domain. Call-ID: 3125316c8a2a1...@112.xxx.xx.xxx. CSeq: 56 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Content-Length: 0. . # U 182.xxx.xx.xxx:5060 - 112.xxx.xx.xxx:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: sip:02xx...@my.domain;tag=9A806878F0882CFC. To:
Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hello Brian, yes we have updated to the latest Fritzbox Firmware. These FritzBoxes are widely spread here in Germany. I know of a SIP provider who has 5 Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in Germany, and they are covering a big stake of in the market. So they generally they work. I tested mine against my Asterisk without problems. But in my Freeswitch environment this is not working, and we have manage to couple of these Boxes. So any help is appreciated. Best regards Peter Brian West schrieb: Besides taking a hammer to it? Have you tried to make sure you have the latest firmware? Try setting the ptime on the fritz to 20ms? I really can't trust a product that has fritz in its name... it might go on the fritz :P pun intended. /b On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote: Any more hints? ___ 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
[Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Hello, anybody has a clue what this message means? [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS What does VETO mean here? Best regards Peter ___ 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] TDM API: CMD: 18 : Operation not supported
Hello, I setup libpri and a sangoma card A108DE, but I cannot dial out. At startup I receive on the D channel TDM API: CMD: 18 : Operation not supported When dialling Libpri debug shows that the numbering plan is fine and that it accepts the screened number, but then it finally hangs up with: Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (e.g. unknown message) (6) ] Does lipri sent any incorrect message here? Protocol is EuroISDN (Q.931/Q.921). Anybody has discovered this already? I am on trunk 14419. See debug and configs below. Best regards Peter Starting FS: 2009-08-03 21:37:18.264829 [DEBUG] zap_io.c:2281 span 1 [d-channel]=[1:16] TDM API: CMD: 18 : Operation not supported 2009-08-03 21:37:18.264915 [INFO] ozmod_wanpipe.c:287 configuring device s1c16 as OpenZAP device 1:16 fd:55 DTMF: none 2009-08-03 21:37:18.264929 [DEBUG] zap_io.c:2281 span 1 [b-channel]=[1:17-31] Dialling: 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 3 to (but not including) 4 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Protocol Discriminator: Q.931 (8) len=14 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Call Ref: len= 2 (reference 5/0x5) (Terminator) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Message type: STATUS (125) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 [08 04 82 e3 98 74] 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Cause (len= 6) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (e.g. unknown message) (6) ] 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Cause data 1: 98 (152, Non-Locking Shift To Codeset 0 IE) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Cause data 2: 74 (116, Redirecting Number IE) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 [14 01 01] 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Initiated (1) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE 8 (cs0, Cause) 2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE 20 (cs0, Call State) 2009-08-03 21:16:19.441416 [ERR] ozmod_libpri.c:88 Received unsolicited status: Info. element nonexist or not implemented openzap.conf [span wanpipe PRI_1] number = 1 trunk_type = E1 b-channel = 1:1-15 d-channel = 1:16 b-channel = 1:17-31 openzap.conf.xml configuration name=openzap.conf description=OpenZAP Configuration settings param name=debug value=10/ /settings libpri_spans span name=PRI_1 param name=node value=cpe/ param name=dp value=national/ param name=l1 value=alaw/ param name=switch value=euroisdn/ param name=dialplan value=XML/ param name=context value=public/ /span /libpri_spans ___ 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
Re: [Freeswitch-users] Sangoma a101
I was asked by telco to specifically use telco, will using Q931 cause any issues for me? Sorry, I do not not understand that question. Best regards Peter Niall Crosby schrieb: Thanks for help. Got my alarms sorted - telco had problem there. As for my Freeswitch problem, chaning to Q931 clears the error message. I was asked by telco to specifically use telco, will using Q931 cause any issues for me? Thanks again, Niall. 2009/7/31 Peter P GMX prometheus...@gmx.net: Firstly I would try get get rid of the sangoma errors. There are 2 errors: Short Circuit: ON Loss of Signal: ON So either the card is not configured correctly or there is a cabling problem. What is your wanpipe1.conf? Second, if euro doesn't work, try Q931. I had success with that some time ago. Best regards Peter Niall Crosby schrieb: Hi, This might be Sangoma config issue, so apologies in advance for posting it here if it is. I am waiting for Sangoma helpdesk to get back to me! But I have a Sangoma a101 and trying to get it working with Freeswitch. Have E1 line coming from telco and everything set up correctly (to my best efforts), however when Freeswitch starts, it says: 2009-07-30 15:17:15.446569 [ERR] mod_openzap.c:1953 Error starting OpenZAP span 1 mode: user dialect: euro error: Also I'm getting the following Sangoma alarms: * w1g1: E1 Alarms (Framer) * ALOS: OFF | LOS: ON RED:ON | AIS: OFF OOF:ON | RAI: OFF * w1g1: E1 Alarms (LIU) * Short Circuit: ON Open Circuit: OFF Loss of Signal: ON Can someone tell me where I should be focusing my attention - on the Sangoma config or the Freeswitch config? Also what state should the Sangoma alarms be in when Freeswitch is NOT running? Should they all clear? Thanks in advance, Niall. ___ 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
Re: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds
Hello Luis,
are you using encrypted TLS instead on SIP on this phone? I experienced
a similar behaviour with 31 seocnds on TLS.
Best regards
Peter
Luis F Urrea schrieb:
Hi all,
I am experiencing a behavior that I cannot clearly understand.
Basically I autocall a few phones into a conference with the
sip_auto_answer set to true, as follows:
extension name=extension-intercom
condition field=destination_number expression=^773$
action application=set
data=conference_auto_outcall_prefix={sip_auto_answer=true}/
action application=conference_set_auto_outcall
data=user/305/
action application=conference_set_auto_outcall
data=user/303/
action application=conference_set_auto_outcall
data=user/201/
action application=conference data=412+flags{endconf|deaf}/
action application=conference data=412 kick all/
/condition
/extension
The conference establishes just fine and everyone can hear just fine.
The strange behavior comes when the person calling to ext 773 hangs
up before 31 seconds have passed, the rest of the phones stay up until
they reach second 31 into the conference.
I am using snom phones and I see the BYE message arriving at the
phones exactly at second 31 after the call establishes.
The conference itself however does not exist after the person calling
773 hangs up (doing conference list on CLI shows NO active conferences).
If the conference stays up more than 31 seconds, then when the person
calling 773 hangs up, the rest of the phones hang up immediately as
desired.
Here's the log for a page that lasts less than 31 seconds:
http://pastebin.freeswitch.org/9773
Here's the log of the phone for a page that lasts less than 31 seconds:
http://pastebin.freeswitch.org/9774
Your inout is appreciated.
Regards,
Luis
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Re: [Freeswitch-users] FsGUI
Thanks, I have found the sources in contrib/jmesquita/fsgui Any recommendatioins how to compile it under Linux? Best regards Peter João Mesquita schrieb: Dear folks, Even tho it might be premature, I would like to already spread the word. Check out FsGUI and feel free give feedback if this is a wanted tool and what direction it should take. Beware that the code is still contrib code and might now be yet mature for production use. http://wiki.freeswitch.org/wiki/Fsgui Thanks, João Mesquita ___ 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
Re: [Freeswitch-users] show channels command with duration - patch included
Well, that's very useful for us in order to have this info in our FS Operator panel. Best regards Peter freeswitch-users@lists.freeswitch.org schrieb: Hi, I usually find it very useful when I can retrieve a list of the currents calls along with durations. I noticed that the 'show channels' format does not include the duration (or the answered timestamp - so that one can extract it from there). So, I made a patch that includes the answered timestamp, the answered timestamp in epoch, and the duration in seconds. Of course these fields remain empty when the call hasn't been answered yet. I don't know if anyone else finds this functionality useful, so I am posting this patch here first (instead of JIRA) in order to get feedback from the users. If many of you (or the maintainers) find it interesting I can then proceed in posting it to JIRA. ___ 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
Re: [Freeswitch-users] Error in default Sofia profile checking
Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP register looks as follows. As you can see, the contact header is there. U 127.0.0.1:5062 - 127.0.0.1:5060 REGISTER sip:127.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 192.168.178.146:5062;rport;branch=z9hG4bKtvnvzdwy. Max-Forwards: 70. To: 8353310 sip:8353...@127.0.0.1. From: 8353310 sip:8353...@127.0.0.1;tag=avpju. Call-ID: ibubkykiithq...@192.168.178.146. CSeq: 5792 REGISTER. Contact: sip:8353...@192.168.178.146:5062;expires=60. Authorization: Digest username=8353310,realm=127.0.0.1,nonce=4bcfe1b0-6f8f-11de-bc32-2dff86a04420,uri=sip:127.0.0.1,response=922690317852a402052da6f74f7196df,algorithm=MD5,cnonce=k9662kmk64,qop=auth,nc=0001. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0.1. Content-Length: 0. . # U 127.0.0.1:5060 - 127.0.0.1:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.178.146:5062;rport=5062;branch=z9hG4bKtvnvzdwy;received=127.0.0.1. From: 8353310 sip:8353...@127.0.0.1;tag=avpju. To: 8353310 sip:8353...@127.0.0.1;tag=4p5K211F33N2c. Call-ID: ibubkykiithq...@192.168.178.146. CSeq: 5792 REGISTER. Contact: sip:8353...@192.168.178.146:5062;expires=60. Date: Mon, 13 Jul 2009 09:26:51 GMT. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12955M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. Can you ngrep your traffic and port your register request? ngrep -d any port 5060 -W byline Best regards Peter velusamy velu schrieb: Dear Peter, I have followed your steps, For me my FS and Twinkle running in separate machine. But, I am still receiving the same error [ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO CONTACT! Please give any suggestions to rectify this error.. Thanks in Advance, Regards, K.Velusamy. On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: I have several Twinkles running against freeswitch on a locally installed machine (FS acts as a SIP/TLS proxy). So in general Twinkle works (on various Ubuntu machines from 7 upto 9 with various Twinkle versions). It must be some kind of setting in Twinkle. E.g. * set the local Twinkle SIP UDP port to 5062 in general settings * Set the right network interface (e.g. eth0) * In the profile do not set the realm * Allow missing contact header on 200 OK Best regards Peter Mathieu Rene schrieb: Chances are the registering UA didnt provide a Contact header (required by rfc3261) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca mailto:mr...@avgs.ca On 11-Jul-09, at 1:23 AM, velusamy velu wrote: Dear Friends, When I register my Softphone(Twinkle) with predefined sofia registration(1000 with password 1234). I have got the following error in FreeSWITCH console. 2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_ register() NO CONTACT! Please help me to solve this problem... Regards, K.Velusamy. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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 http://www.freeswitch.org/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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 http://www.freeswitch.org/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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 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
Re: [Freeswitch-users] Error in default Sofia profile checking
I have several Twinkles running against freeswitch on a locally installed machine (FS acts as a SIP/TLS proxy). So in general Twinkle works (on various Ubuntu machines from 7 upto 9 with various Twinkle versions). It must be some kind of setting in Twinkle. E.g. * set the local Twinkle SIP UDP port to 5062 in general settings * Set the right network interface (e.g. eth0) * In the profile do not set the realm * Allow missing contact header on 200 OK Best regards Peter Mathieu Rene schrieb: Chances are the registering UA didnt provide a Contact header (required by rfc3261) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 11-Jul-09, at 1:23 AM, velusamy velu wrote: Dear Friends, When I register my Softphone(Twinkle) with predefined sofia registration(1000 with password 1234). I have got the following error in FreeSWITCH console. 2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_ register() NO CONTACT! Please help me to solve this problem... Regards, K.Velusamy. ___ 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 ___ 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
Re: [Freeswitch-users] pocketsphinx
Hello Helmut, I looked at these dic files. Their content (look at all the qq's) is quite different from the dic files supplied with freeswitch pocketsphinx. As I remember the CMU dict file format has changed in April 2008. Best regards Peter Helmut Kuper schrieb: Hi, I try to change pocketsphinx's grammar from default (english) to german. I found this archive (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), which contains similar files like those which can be found in grammar/model/communicator directory. Unfortunately FS crashed without writing a core file nor logfile enries as soon as as pizza demo trys to detect speech. Any Ideas? Maybe someone has already working grammar/model files for german language? regards helmut ___ 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
Re: [Freeswitch-users] pocketsphinx
Hello Helmut, I looked at these dic files. Their content (look at all the qq's) is quite different from the dic files supplied with freeswitch pocketsphinx. As I remember the CMU dict file format has changed in April 2008. Maybe there is a converter somewhere? I was thinking of just enhancing the current dict file for some german words I need, but did not test it so far. This should be possible without modifying the underlying grammar. http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary I would love to hear when you have had any progress on this. Best regards Peter Helmut Kuper schrieb: Hi, I try to change pocketsphinx's grammar from default (english) to german. I found this archive (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), which contains similar files like those which can be found in grammar/model/communicator directory. Unfortunately FS crashed without writing a core file nor logfile enries as soon as as pizza demo trys to detect speech. Any Ideas? Maybe someone has already working grammar/model files for german language? regards helmut ___ 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
[Freeswitch-users] Sangoma 108 and libpri problems - only distortion sound
Hello,
I installed a Sangoma A108 with openzap and libpri. Signalling (E1)
works sometimes (inbound and outbound calls are connected) but not always.
Sound is just distortion but connection is stable.
2 questions:
1) What is the best way to go with Sangoma? OpenZAP with libpri or
without libpri? (I remember there are some timer problems in openzap
when not using libpri but this might have changed) However Sangoma
recommends OpenZAP on their wiki.
2) What might cause the distortion? I crosschecked the config files and
had a look at the interrupts (2k/sec). Seems to be ok. ACPI and APIC is
turned on.
Freeswitch starts successfully with all the channels enabled. oz dump
shows D-Channel up.
oz libpri debug 1 all shows debugging messages with no special warnings.
Best regards
Peter
Some confs for 1 Channel:
Wanpipe1.conf
[devices]
wanpipe1 = WAN_AFT_TE1, Comment
[interfaces]
w1g1 = wanpipe1, , TDM_VOICE_API, Comment
[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 4
PCIBUS = 11
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= CRC4
FE_LINE = 1
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 16
TDMV_HW_DTMF= YES
TDMV_HW_FAX_DETECT = YES
[w1g1]
ACTIVE_CH = ALL
TDMV_HWEC = YES
MTU = 80
openzap.conf
[span wanpipe PRI_1]
name = OpenZAP
number = 1
trunk_type = e1
b-channel = 1:1-15
d-channel = 1:16
b-channel = 1:17-31
openzap.conf.xml
configuration name=openzap.conf description=OpenZAP Configuration
settings
param name=debug value=1/
param name=hold-music value=$${moh_uri}/
!--param name=enable-analog-option value=call-swap/--
!--param name=enable-analog-option value=3-way/--
/settings
libpri_spans
span id=1 name=PRI_1
param name=node value=cpe/
param name=switch value=euroisdn/
param name=dialplan value=XML/
param name=context value=default/
param name=q921loglevel value=debug/
param name=q931loglevel value=debug/
param name=dialect value=q931/
param name=l1 value=alaw/
/span
/libpri_spans
/configuration
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[Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?
Hello, I have the following problem: Every call stops after 30 seconds when TLS is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. The phones are behind NAT. So I expect, that every 30 seconds an Options request is sent. Wiresharking the traffic I can see * that there are ongoing UDP packets. * Then a TSLv1 packet ist sent from FS to the Phone. * This is acknowleged by the phone * Next the phone send another UDP packet to the same FS port as before * Then the Phone receives an ICMP request that the FS port is closed. Anybody has a clue about this? Best regards Peter ___ 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
Re: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?
Some additions: TLS/RTP instead of SRTP does also not work. There are no logs on the debug console except the message that the call is being terminated 2009-07-02 12:06:45.252177 [DEBUG] sofia.c:3100 Channel sofia/internal/835...@sip.mydomain.de entering state [terminating][0] and later cause: NORMAL_UNSPECIFIED Best regards Peter Peter P GMX schrieb: Hello, I have the following problem: Every call stops after 30 seconds when TLS is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. The phones are behind NAT. So I expect, that every 30 seconds an Options request is sent. Wiresharking the traffic I can see * that there are ongoing UDP packets. * Then a TSLv1 packet ist sent from FS to the Phone. * This is acknowleged by the phone * Next the phone send another UDP packet to the same FS port as before * Then the Phone receives an ICMP request that the FS port is closed. Anybody has a clue about this? Best regards Peter ___ 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
Re: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?
Hello Brian, ok, I got it. Any other idea why the UDP port is closed after the TLS packet? Best regards Peter Brian West schrieb: If its TLS you don't need options packets in the first place. Your client should do the keep alive NOT FreeSWITCH. TLS is over TCP and Options over UDP... doesn't make much sense. /b On Jul 2, 2009, at 6:11 AM, Peter P GMX wrote: Hello, I have the following problem: Every call stops after 30 seconds when TLS is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. The phones are behind NAT. So I expect, that every 30 seconds an Options request is sent. Wiresharking the traffic I can see * that there are ongoing UDP packets. * Then a TSLv1 packet ist sent from FS to the Phone. * This is acknowleged by the phone * Next the phone send another UDP packet to the same FS port as before * Then the Phone receives an ICMP request that the FS port is closed. Anybody has a clue about this? Best regards Peter ___ 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
Re: [Freeswitch-users] How to know my gateway registering is successed??
or simply sofia status for all gateways Jason White schrieb: Brad Tuan brad.t...@gmail.com wrote: As title ,Does FS keep the status of gateways?? sofia status gateway gateway-name ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Ray, I do use event socket and it pushes me a link on the website whenever a call for this agent comes in. It's just a matter of visibility. The agent may still finish his old workflow and is still entering data. When a call comes in then and he picks up the phone, the data he just entered is gone away. So I would like the web app to drive answering the call. It gives a better visibility about what he is doing to the callcenter agent. Best regards Peter Raymond Chandler schrieb: Peter P GMX wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. is there any reason you don't make your web app listen to event socket or event sink to catch the answer event and start the workflow? then you just need to answer the call on the softphone and the webapp should automatically start the workflow. -Ray ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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
Re: [Freeswitch-users] Zoiper reject freeswitch calls
May this help also: I just tried current Zoiper with TLS. Outbound is working, inbound not. Zoiper registeres with the following contact info: 7233213 sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS When a call comes in, Zoiper rings once and then hangs up. It shows service or option not implemented in the Zoiper log. My snom phones with the same parameters in the same network (they are all nated) register differently 723323 sip:723...@192.168.178.143:2059;transport=tls;line=4xbyd8h3;fs_nat=yes;fs_path=sip%3A723323%40217.xx.xx.xxx%3A2059%3Btransport%3Dtls%3Bline%3D4xbyd8h3 My FS logs show for an incoming call to Zoiper: 7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) Running State Change CS_CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] switch_core_state_machine.c:502 (sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS) State CONSUME_MEDIA 2009-06-16 14:50:16.336881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [calling][0] 2009-06-16 14:50:16.340881 [DEBUG] sofia.c:3100 Channel sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS entering state [terminated][415] 2009-06-16 14:50:16.340881 [NOTICE] sofia.c:3660 Hangup sofia/internal/sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Its seems that something with the codecs fails here, although I have enabled all codecs in Zoiper and FS offers alaw. Best regards Peter Brian West schrieb: Ok i'll have to se what I can do about reproducing this issue now that I have more info on how its happening. /b On Jun 16, 2009, at 7:40 AM, dujinfang wrote: Almost caught you on IRC Mike. Our server is in a NAT'd network and all agents registered in the same LAN. I can remotely register by using the public IP and the contact string is right. Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE. User: 6...@192.168.1.16 mailto:6...@192.168.1.16 Contact:user sip:6...@210.73.8.180:5090;rinstance=9fc589bbc407518f Agent: Zoiper rev.1809 So it's like only happens on our LAN and where there's a fs_path present. Just curious, why agents registered on a local LAN has param fs_nat=yes; (default internal profile, port 5060) ? Seems our time doesn't match, I'm generally available in office 9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow. Thank you. ___ 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
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
It mainly works now by uuid_transfer the following way via event socket. uuid_setvar unique_id sip_invite_params intercom=true uuid_setvar unique_id sip_auto_answer true uuid_transfer unique_id 1000 XML default so the call is transferred from 1000 to 1000. What happens: 1) If I disable intercom on the Snom phone, the phone rings, stops ringing and rings again (ok) 1) If I enable intercom on the Snom phone, the phone rings, stops ringing and hangs up (not ok) So I do not get the Snom to pick up the call in intercom mode. The last invite is: INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0 Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib Max-Forwards: 68 From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls Call-Info: sip:217.xx.xx.xxx;answer-after=0 The intercom part is there and the Call-Info line with answer-after also. The phone answers with SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF To: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true;tag=71rskygkr2 Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e CSeq: 116467629 INVITE Contact: sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1 WWW-Authenticate: Digest realm=sip2.mycompany.de, nonce=2ee26efe6ab27f88, algorithm=MD5 Content-Length: 0 and hangs up. Anybody know how to solve this Snom intercom issue? Best regards Peter Michael Jerris schrieb: The transfer should work but it sounds like offhook agents is what your really trying to accomplish so I would go with brian's suggestion. On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I want the phone be ringing, just for acoustical feedback reasons. But what if I * transfer it to the same user destination again (now with intercom enabled), will this work? * transfer it to park and then transfer it to the same destination again (now with intercom enabled) Best regards Peter Michael Jerris schrieb: The only way I can think to do this today would be to cancel the call and re send with the intercom headers for a phone that supports it. It may be possible to send a reinvite with autoanswer headers but I doubt that would work, all you could do is try making code to do it it a sipp or sipsak scenario and test it. A better aproach might be to answer the call normally and detect that to start your web workflow or not really ring the phone, just the web app and deliver the call with autoanswer when the button is hit in the web ui. Mike On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, this is too easy :-). This is for a small callcenter app and I only want the user to pickup the call once (to accept the call in X-Lite (or a Snom phone) and to start the workflow on the web application). I do not want him to accept the call on the phone and then on the Web app. Best regards Peter Brian West schrieb: click on the AA button? :) /b On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote: What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? ___ 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 ___ 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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http
Re: [Freeswitch-users] Force SIP UA to pick up call during ringing?
Thanks Michael,
I have disabled it now.
I finally got it to work, (sip_h_Call-Info=sip:$${domain};answer-after=0)
but the behaviour was not as desired, as I didn't manage the phone to
pick up the call on the headset. It will only have the speaker enabled.
So I will have to go a different way with parking the call and then
forward it.
Best regards
Peter
Michael Jerris schrieb:
uuid_setvar unique_id sip_invite_params intercom=true should be
unnecessary.
Mike
On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
It mainly works now by uuid_transfer the following way via event
socket.
uuid_setvar unique_id sip_invite_params intercom=true
uuid_setvar unique_id sip_auto_answer true
uuid_transfer unique_id 1000 XML default
so the call is transferred from 1000 to 1000.
What happens:
1) If I disable intercom on the Snom phone, the phone rings, stops
ringing and rings again (ok)
1) If I enable intercom on the Snom phone, the phone rings, stops
ringing and hangs up (not ok)
So I do not get the Snom to pick up the call in intercom mode.
The last invite is:
INVITE sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib
SIP/2.0
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
Route: sip:1...@217.24.11.189:2752;transport=tls;line=er6kxnib
Max-Forwards: 68
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact: sip:mod_so...@217.xx.xx.xxx:5061;transport=tls
Call-Info: sip:217.xx.xx.xxx;answer-after=0
The intercom part is there and the Call-Info line with answer-after
also.
The phone answers with
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
From: Peter FS sip:723...@217.xx.xx.xxx;tag=9eQ8rjQy533HF
To:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true
;tag=71rskygkr2
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact:
sip:1...@192.168.178.50:2752;transport=tls;line=er6kxnib;reg-id=1
WWW-Authenticate: Digest realm=sip2.mycompany.de,
nonce=2ee26efe6ab27f88, algorithm=MD5
Content-Length: 0
and hangs up.
Anybody know how to solve this Snom intercom issue?
Best regards
Peter
Michael Jerris schrieb:
The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's
suggestion.
On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Michael,
I want the phone be ringing, just for acoustical feedback reasons.
But what if I
* transfer it to the same user destination again (now with
intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with intercom enabled)
Best regards
Peter
Michael Jerris schrieb:
The only way I can think to do this today would be to cancel the
call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it
it
a sipp or sipsak scenario and test it. A better aproach might be
to
answer the call normally and detect that to start your web workflow
or
not really ring the phone, just the web app and deliver the call
with
autoanswer when the button is hit in the web ui.
Mike
On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net
wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to
pickup
the call once (to accept the call in X-Lite (or a Snom phone)
and to
start the workflow on the web application). I do not want him to
accept
the call on the phone and then on the Web app.
Best regards
Peter
Brian West schrieb:
click on the AA button? :)
/b
On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
What is the best way to have this done? Move the call to park
and
then
retransfer again with intercom, or is there a better solution?
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[Freeswitch-users] Force SIP UA to pick up call during ringing?
I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter ___ 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] xmpp
I saw that xmpp is supported in Fresswitch. See wiki: http://wiki.freeswitch.org/wiki/Mod_xmpp_event Has anybody already set this up? I have found no mod_xmpp neither in my mod directory nor in the source? There was also a question: Q: Is it possible to send commands to fs via xmpp? Answer: Yes. Anybody knows what can be done here and how to do this? Best regards Peter ___ 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
Re: [Freeswitch-users] Error sending mail
I have a problem where FS gives a core file when an voicemail email shall be sent via exim. I am on 13438. No entry in debug log in FS. No entry in exim log. Best regards Peter Jason White schrieb: Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote: 1.0.4pre8 It works for me with revision 13501. Mine is later than yours. Try upgrading. ___ 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
Re: [Freeswitch-users] Error sending mail
JIRA opened: *FSCORE-375 http://jira.freeswitch.org/browse/FSCORE-375* Brian West schrieb: Please Open a JIRA ASAP. We are working to get 1.0.4 out and these are the types of issues that should have been reported weeks ago if they were happening. /b On May 30, 2009, at 6:27 AM, Peter P GMX wrote: I have a problem where FS gives a core file when an voicemail email shall be sent via exim. I am on 13438. No entry in debug log in FS. No entry in exim log. Best regards Peter Jason White schrieb: Luis M. Zuccolo luismzucc...@yahoo.com.ar mailto:luismzucc...@yahoo.com.ar wrote: 1.0.4pre8 Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ 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
Re: [Freeswitch-users] The calls are dropped during register
And mine with the same behaviour on Linux. Best regards Peter Diego Toro schrieb: Hi, my job with FS has been on Windows. Diego --- On *Thu, 5/28/09, Brian West /br...@freeswitch.org/* wrote: From: Brian West br...@freeswitch.org Subject: Re: [Freeswitch-users] The calls are dropped during register To: freeswitch-users@lists.freeswitch.org Date: Thursday, May 28, 2009, 8:22 PM Anything on linux? /b On May 28, 2009, at 8:19 PM, Diego Toro wrote: thank you Brian, my notes on http://jira.freeswitch.org/browse/SFSIP-143, I have hardware avaible for testing Diego Brian West br...@freeswitch.org http://us.mc335.mail.yahoo.com/mc/compose?to=br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ -Inline Attachment Follows- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 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 ___ 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] Origionate a call via event_socket. relation between job_uuid and uuid
I want to do the following: Originate a call via event_socket, I get back a job_uuid. Then I want to control the call when it's established (2 call legs). Scanning the variables of the 2 call legs I currentyl cannot see any relation between the job_uuid and the uuid of the resulting call legs. I may set a variable with my own unique id while originating a call, but finding the calls later on and dumping the variables fo all channels is very time consuming in terms of CPU. A workaround I tried, is to set caller-id or caller-id-number with a unique id. This works, but has the known side effects of not having a valid caller-id or caller-id-number. So my question is: Has anybody an idea how to build a relationship between job_uuid and the resulting call legs which does not require dumping the variables of all channels? Best regards Peter ___ 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
Re: [Freeswitch-users] Origionate a call via event_socket. relation between job_uuid and uuid
Hello Thanks for your hints,
I now added {initiator_uuid=my_uuid} prefix to the dialstring.
Then I catch the channel_answer event, get this variable_initiator_uuid
and pass it to the application. This works like a charm.
Thanks to all.
Best regards
Peter
Anthony Minessale schrieb:
Here are 3 ways:
1) subscribe to the BACKGROUND_JOB event and find the one with the
same job-uuid
then the body of that message is the output from your backgrounded
FSAPI call which in the case
of an originate will contain the uuid of the actual channel.
2) You can do as suggested and add {myvar=myval} prefix to the
dialstring and look for
myvar in the channel_originate event.
3) Finally you can choose the uuid in advance providing it's actually
unique using:
{origination_uuid=XYZ}
You can use your own code to generate uuid (make sure they are
unique) or
ask the core to give you a new uuid with the create_uuid FSAPI call.
On Wed, May 27, 2009 at 4:46 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
I want to do the following:
Originate a call via event_socket, I get back a job_uuid. Then I
want to
control the call when it's established (2 call legs).
Scanning the variables of the 2 call legs I currentyl cannot see any
relation between the job_uuid and the uuid of the resulting call legs.
I may set a variable with my own unique id while originating a
call, but
finding the calls later on and dumping the variables fo all
channels is
very time consuming in terms of CPU.
A workaround I tried, is to set caller-id or caller-id-number with a
unique id. This works, but has the known side effects of not having a
valid caller-id or caller-id-number.
So my question is: Has anybody an idea how to build a relationship
between job_uuid and the resulting call legs which does not require
dumping the variables of all channels?
Best regards
Peter
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Re: [Freeswitch-users] Calls drop at 30 seconds
I had a similar behaviour with dropped calls. After I changed the firewall on the FS machine it worked. In my case some ports on the FS machine were not open for outbound traffic (inbound were ok). Check SIP, TLS, RTP, STUN, DNS ports. Best regards Peter FERNANDO VILLARROEL schrieb: Hi Diego, The softphones are in different machines (Softphone 1 Win XP and Softphone 2 in other win XP), i have ringback, but no audio and call death at 30 seconds. Fernando --- On Mon, 5/25/09, Diego Viola diego.vi...@gmail.com wrote: From: Diego Viola diego.vi...@gmail.com Subject: Re: [Freeswitch-users] Calls drop at 30 seconds To: freeswitch-users@lists.freeswitch.org Date: Monday, May 25, 2009, 5:13 PM I had the same issue before, and it was a LAN problem, make sure your network is configured properly. Are you running the softphones and FS on the same machine? Diego On Mon, May 25, 2009 at 7:44 PM, FERNANDO VILLARROEL fvillarr...@yahoo.com wrote: Hi, I have 2 softphones (101 and 102) logged to my FS in a LAN, but the calls drop at 30 seconds: 2009-04-29 22:44:12 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/admin/1...@192.168.1.150 entering state [terminating][0] 2009-04-29 22:44:12 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/admin/1...@192.168.1.150 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/1...@192.168.1.150 [KILL] 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/1...@192.168.1.150 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/1...@192.168.1.150] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 audio_bridge_thread() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/admin/102 receive message [UNBRIDGE] 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:630 switch_core_session_perform_receive_message() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:452 audio_bridge_thread() BRIDGE THREAD DONE [sofia/admin/102] 2009-04-29 22:44:12 [DEBUG] switch_ivr_bridge.c:456 audio_bridge_thread() Send signal sofia/admin/1...@192.168.1.150 [BREAK] 2009-04-29 22:44:12 [NOTICE] switch_ivr_bridge.c:505 audio_bridge_on_exchange_media() Hangup sofia/admin/102 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-04-29 22:44:12 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/admin/102 [KILL] 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:493 switch_core_session_run() (sofia/admin/102) State EXCHANGE_MEDIA going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/102) Running State Change CS_HANGUP 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/102) State HANGUP 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/admin/102 hanging up, cause: NORMAL_CLEARING 2009-04-29 22:44:12 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending BYE to sofia/admin/102 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/admin/102 Standard HANGUP, cause: NORMAL_CLEARING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/admin/102) State HANGUP going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/admin/102) State Change CS_HANGUP - CS_REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/admin/102 [BREAK] 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/102) Running State Change CS_REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/admin/102) State REPORTING 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/admin/1...@192.168.1.150) State EXECUTE going to sleep 2009-04-29 22:44:12 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/admin/1...@192.168.1.150) Running State Change CS_HANGUP 2009-04-29 22:44:12 [DEBUG]
[Freeswitch-users] uuid_chat
Hello, today I tried uuid_chat via event socket. A simple chat application works: bgapi chat sip|age...@fqdn|age...@fqdn|Message. uuid_chat uuid however returned +OK, but nothing happens. Neither is there a debug line on the console, nor a SIP (in my case TLS) message is sent to the UA. Has anybody successfully tried this command and has some additional hints? Is there any further configuration needed? Best regards Peter ___ 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
Re: [Freeswitch-users] FS in Amazon EC2 for production?
We have used FS on ec2 for testing purposes only. It was ok. We havn't done any performance test though. Best regards Peter Ing. Edwin Villarreal schrieb: Hello my friends. Has anyone used the EC2 for production? Tests? I’m wondering if it would be “better” to have a FS system in the cloud for carrier-to-carrier connections. Any ideas will be appreciated Thanks 2 u all *Edwin Villarreal* World Net Commerce SA CV WNC Telecom ___ 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
Re: [Freeswitch-users] Cool names for my VoIP company
Just a side notice about how to name a company. If you use a descriptive name e.g. GlobalSIP as sugested before, it may be difficult to register this name later on as a brand name when your company becomes successful. At least here in Europe it is not possible to register a brand name when the name itself describes the business or the techique used. Thats the reason why big companies nowadays use these strange names like e.g. ABALA, which seem to not make any sense at all at a first glance. But these names can eaysily be registered as brand names. Best regards Peter Ognjen Seslija schrieb: I vote for viotel. Regards, Ognjen On Fri, May 22, 2009 at 6:26 AM, Diego Viola diego.vi...@gmail.com mailto:diego.vi...@gmail.com wrote: Hey guys, I'm about to start my own ITSP with FreeSWITCH, and I'm looking some cool names for my VoIP company, if you know some please tell me :) Diego ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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 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 ___ 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
Re: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy
This is also interesting for me, as I love freeswitch, and maintaining a single platform is easier, than handling various different ones. In the past years I did a couple of projects with OpenSER /openSIPS. These projects comprised: * registrar for the SIP user agents * handle invite messages (+ ringing, bye, ok, etc also of course) between registered user agents and user agents at external domains * rtp payload was a bit different from usual VoIP traffic (video parts, application sharing, file downloads etc.), but SDP was fine according to RFC, and OpenSER mediaproxy worked also * handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS) * The number of messages to handle was not that much (some thousand subs). For my understanding this should also be possible with Freeswitch with bypass_media. Right? Best regards Peter Ognjen Seslija schrieb: Hello, FS by design is B2BUA, and it cannot route INVITEs and other SIP methods. It can however, bridge a-leg to b-leg with or w/o media and doing plenty other cool stuff much better than commercial projects. I suggest joining us on irc to detail your setup so we can help you. Regards, Ognjen (sekil on #freeswitch). On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar) sridh...@alcatel-lucent.com mailto:sridh...@alcatel-lucent.com wrote: Hi all, I want to use FreeSWITCH as a SIP transparent proxy in session border controller application. Please let me know the changes in configuration files required to achieve this behaviour Thanks very much for the help. Regards, Sridhar ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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 ___ 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
Re: [Freeswitch-users] silly (?) questions
Hello Jean-Yves,
did you ever try a call-trough? (a person dials in (1234567, see below)
types the target number as DTMF and gets connected to this number?
A basic dialplan can be like this:
extension name=Dialthru
condition field=destination_number expression=^(1234567)$
action application=play_and_get_digits data=5 25 3 7000 #
ivr/8000/ivr-enter_ext.wav voicemail/8000/vm-that_was_an_invalid_ext.wav
foobar \d+/
action application=transfer data=${foobar} XML default/
/condition
/extension
For \d+ you may define your regular expression, which numbers you
would accept. Also you may try to redirect into the dialplan again after
the number is entered (instead of directly transferring the call).
Best regards
Peter
Jean-Yves F. Barbier schrieb:
Hi list,
I just discovered FS (practiced a bit * 2 years ago, but too much unstable)
and find it cool, NOT CPU greedy and (almost) working ouf of the web.
I'd like to know if star codes (such as *98) are normalized or not?
(and if so, where I could find a list)
Also, as I don't use very much my phone and mostly don't pay for it (I
live in france and got unlimited free call for 70-80 countries, and
my phone is actually plugged in my ADSL box) I'd like to leave access
for other people through something like DUNDi (that I don't really
know.)
BUT not everything is free (i.e.: cellular phones calls cost €0.22 @
connection + €0.22/min); thus I must forbid this kind of calls.
Does anybody have realized that, because I need a good template?
Thanks
JY
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Re: [Freeswitch-users] Call For Participants: Lightning Talks at ClueCon 2009
Hello Michael, I will have to fly in from Germany. So if there's 90% chance to speak, fine. If there's only 50% chance, ??? How would you rate the chance? Best regards Peter Michael S Collins schrieb: We are pretty much booked solid as we've got some unconfirmed speakers we haven't posted yet. I'm redoing the schedule and will have an updated one out this next week. One thing that we really need is backup speakers. Our experience is that there are always people who have emergencies and can't make it. Would you be willing to be one of our backups? There is a pretty good chance that you would speak but we won't know exactly which day or time. Please let me know what you think. -MC Sent from my iPhone On May 16, 2009, at 6:14 AM, Peter P GMX prometheus...@gmx.net wrote: Hello Michael, I see that there are still some time slots available on 6th of Aug. I am thinking of doing a presentation on an application server and Web GUI for Fresswitch we have developed. Is it still possible to register for a full speaker slot? Best regards Peter Michael Collins schrieb: *ClueCon 2009 is coming soon!* We are interested in your thoughts on subjects for lighting talks. We would love to have a number of 5-10 minute presentations by members of the community. If you would like to give a talk, or just have an idea for a talk, please let us know. How do lightning talks work? Quite simply, the presenter has just a few minutes to speak on a particular subject, usually no more than 10 minutes. He or she will deliver the information rapidly, which means keeping the presentation focused tightly on the subject being discussed. Lightning talks usually do not have enough time for audience QA. However, ClueCon has a long lunch period that is designed to allow attendees plenty of time to interact. Those are perfect times to discuss lightning talks or any other presentations. Those who give presentations enjoy interacting with other attendees in a relaxed atmosphere during lunch or in the evening at dinner. If you haven't already registered for ClueCon 2009 then please call us at 877.742.CLUE right away and we will complete your registration. Also, don't forget that expedia.com http://expedia.com has some nice hotel deals for the Wyndham Chicago. Book your room today! We look forward to hearing from you and seeing you all at ClueCon in Chicago. -Michael http://www.cluecon.com 877.742.CLUE --- - ___ 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 ___ 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
Re: [Freeswitch-users] Call For Participants: Lightning Talks at ClueCon 2009
Hello Michael, I see that there are still some time slots available on 6th of Aug. I am thinking of doing a presentation on an application server and Web GUI for Fresswitch we have developed. Is it still possible to register for a full speaker slot? Best regards Peter Michael Collins schrieb: *ClueCon 2009 is coming soon!* We are interested in your thoughts on subjects for lighting talks. We would love to have a number of 5-10 minute presentations by members of the community. If you would like to give a talk, or just have an idea for a talk, please let us know. How do lightning talks work? Quite simply, the presenter has just a few minutes to speak on a particular subject, usually no more than 10 minutes. He or she will deliver the information rapidly, which means keeping the presentation focused tightly on the subject being discussed. Lightning talks usually do not have enough time for audience QA. However, ClueCon has a long lunch period that is designed to allow attendees plenty of time to interact. Those are perfect times to discuss lightning talks or any other presentations. Those who give presentations enjoy interacting with other attendees in a relaxed atmosphere during lunch or in the evening at dinner. If you haven't already registered for ClueCon 2009 then please call us at 877.742.CLUE right away and we will complete your registration. Also, don't forget that expedia.com http://expedia.com has some nice hotel deals for the Wyndham Chicago. Book your room today! We look forward to hearing from you and seeing you all at ClueCon in Chicago. -Michael http://www.cluecon.com 877.742.CLUE ___ 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
Re: [Freeswitch-users] pocketsphinx and event socket
Hello Michael, now some time later I did another try with the latest trunk. The problem were the grammar files fr the pizza demo. The old ones didn't work anymore with ne tnew version of pocketsphinx. Now with the new grammar files it works. I have updated the wiki accordingly. Best regards Peter Michael Collins schrieb: On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX prometheus...@gmx.net wrote: Hello Brian, concerning Well you should use ESL then ;) I simply do not understand what you mean by this. Is it sarcastic? Am I asking stupid questions? ESL = Event Socket Library. It is an abstraction layer to make interacting with the FS event socket a little easier. Look in the source directory under libs/esl and you'll see all sorts of stuff. Also check out the new-but-growing ESL wiki page: http://wiki.freeswitch.org/wiki/Esl -MC ___ 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
Re: [Freeswitch-users] Audi record using uuid_record
I record them to file.wav and they play perfectly. I think it's recorded in a raw-format here. See: http://www.nabble.com/Recording-ULAW-files-td21587474.html Best regards Peter Peter Olsson schrieb: Hello again, I also have a problem when I try to record messages. I record to .PCMA-files, and the file is created perfectly. But it’s just distorted audio in it. It sounds to me that there might be a codec issue. The media stream is PCMA all the way from the phone to FreeSWITCH, and to start recording I simply call “uuid_record UUID start c:\test.PCMA”. According to the docs the file should automatically be recorded as PCMA when the file is named .PCMA. Any ideas what I can be doing wrong? Regards, Peter ___ 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
[Freeswitch-users] Double Re-Register problem
Hello, I habe the following problem when re-registering to an external SIP provider during a call which results in immediate call-hangups. - FS re-registers with nonce - 2ms later FS re-registers without nonce - external SIP provider asks for credentials - FS re-registers with nonce - External provider hangs up call I think the external equipment (Huawei) gets his messages into disorder and then hangs up. My question is: How can I force FS to only register once (without nonce)? As said, FS tries to register twice within 2 msecs without receiving an answer in between. FS is on a public IP, so there are no NAT problems expected (I can see that until the re-register takes place, media is passed in both directions). Best regards Peter See log: U 2009/05/07 15:04:37.441636 217.xxx.xxx.190:5080 - 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK43088Ha5QpZBN. Max-Forwards: 70. From: sip:0123456...@sip.provider.de;transport=udp;tag=jSFF9XmFZ50pp. To: sip:0123456...@sip.provider.de;transport=udp. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732262 REGISTER. Contact: sip:gw+xxx_01...@217.xxx.xxx.190:5080;transport=udp. Expires: 0. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Authorization: Digest username=0123456779, realm=provider.de, nonce=4a02dc5ba90f927c74161f89e7550138b93f12cc, cnonce=y8SXDLWpEiyNPQAekEzDTg, algorithm=MD5, uri=sip:sip.provider.de;transport=udp, response=1d44f64eec5a5b38b44e398dea201a08, qop=auth, nc=0002. Content-Length: 0. . # U 2009/05/07 15:04:37.443395 217.xxx.xxx.190:5080 - 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK5ct1aDU8mZNyg. Max-Forwards: 70. From: sip:0123456...@sip.provider.de;transport=udp;tag=t1SNQpUB8cKcK. To: sip:0123456...@sip.provider.de;transport=udp. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732402 REGISTER. Contact: sip:gw+xxx_01...@217.xxx.xxx.190:5080;transport=udp. Expires: 60. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Content-Length: 0. . # U 2009/05/07 15:04:37.466691 213.xxx.xxx.2:5060 - 217.xxx.xxx.190:5080 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;branch=z9hG4bK5ct1aDU8mZNyg;rport=5080. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. From: sip:0123456...@sip.provider.de;transport=udp;tag=t1SNQpUB8cKcK. To: sip:0123456...@sip.provider.de;transport=udp;tag=702dbe10. CSeq: 114732402 REGISTER. Server: SIP Router. WWW-Authenticate: Digest realm=provider.de,nonce=4a02dd789a25b67f29ba21f65429d13c4bbc2ded,qop=auth. Content-Length: 0. . # U 2009/05/07 15:04:37.467211 217.xxx.xxx.190:5080 - 213.xxx.xxx.2:5060 REGISTER sip:sip.provider.de;transport=udp SIP/2.0. Via: SIP/2.0/UDP 217.xxx.xxx.190:5080;rport;branch=z9hG4bK6NKtc8Bcj8BHc. Max-Forwards: 70. From: sip:0123456...@sip.provider.de;transport=udp;tag=t1SNQpUB8cKcK. To: sip:0123456...@sip.provider.de;transport=udp. Call-ID: a0364af0-3b05-11de-bc92-f9fefd954b7b. CSeq: 114732403 REGISTER. Contact: sip:gw+xxx_01...@217.xxx.xxx.190:5080;transport=udp. Expires: 60. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13231M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Authorization: Digest username=0123456779, realm=provider.de, nonce=4a02dd789a25b67f29ba21f65429d13c4bbc2ded, cnonce=dbAgB7WqEiyNPQAekEzDTg, algorithm=MD5, uri=sip:sip.provider.de;transport=udp, response=6a55b27caec6b06bd9da707e7b24d82b, qop=auth, nc=0001. Content-Length: 0. . # U 2009/05/07 15:04:37.470935 213.xxx.xxx.2:5060 - 217.xxx.xxx.190:5080 BYE sip:gw+xxx_01...@217.xxx.xxx.190:5080;transport=udp SIP/2.0. Via: SIP/2.0/UDP 213.xxx.xxx.2:5060;branch=z9hG4bK400739ec3ad9e5bbd7f5edccf. Call-ID: d0e38021-b5a9-122c-3d8d-001e904cc34e. From: sip:0049987654...@sip.provider.de;tag=31de8a21. To: unknownsip:0123456...@sip.provider.de;transport=udp;tag=K287aS5jveQ9H. CSeq: 1 BYE. Max-Forwards: 70. Content-Length: 0. ___ 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
Re: [Freeswitch-users] SRTP Error auth check failed
Hello Helmut, I also have problems with my Snom300s and Snom320s and G711 and SRTP. They may be related to this problem, but I am not sure. The phones disconnect the media stream after a while (2..10 minutes) because the Snom media port is blocked all of a sudden. I have opened a bug report at Snom [Ticket#200904081131]. Best regards Peter Helmut Kuper schrieb: Hello, today a colleague of mine told me that sometimes calls were disconnected without any obvious reasons. In FS's log I found this: 2009-05-07 15:52:22 [ERR] switch_rtp.c:1656 rtp_common_read() Error: SRTP unprotect failed with code 7 (auth check failed) I scanned my recent FS log files for that message and found that this error accours a few times a week. I use Snom Phones all with G722 and SRTP. Any ideas what this could be caused by? regards helmut ___ 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
[Freeswitch-users] conf-is-unlocked.wav missing
Hello, I tried conferencing for FS und tried to lock/unlock conferences. While conf-is-locked.wav was played, conf-is-unlocked.wav was missing in the file system. Any idea where I can download this? Best regards Peter ___ 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
