June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

show caller info
by Kozak Vladimir 18 Jun '10

18 Jun '10

hi everybody. I have one problem. I need to show in phone caller number without IP address or caller number + domain without IP address. I use api originate command with origination_caller_id_name and origination_caller_id_number parameters. bgapi originate {origination_caller_id_name=125@123.12.13.14,origination_caller_id_number=123@123.12.13.14}[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/internal/sip:1009@172.26.10.65:61802;rinstance=437cf350c3a4546f &park() FS cuts value of origination_caller_id_name parameter (delete specified ip) - it's ok. And FS deletes specified ip from value of origination_caller_id_number parameter and adds FS-IP. how else can I show caller info? how can I show caller domain?

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Re : How to enable mod_skypopen without sound card
by Jérôme M. 18 Jun '10

18 Jun '10

On Windows, it exists Virtual Audio Cable to emulate a sound card, here is the website : http://software.muzychenko.net/eng/vac.html But it's not free... 2010/6/17 <freeswitch-users-request(a)lists.freeswitch.org> > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. How to enable mod_skypopen without sound card (k xd) > 2. Aastra SCA Woes, Any Updates? (Wes Pearce) > 3. Re: How to enable mod_skypopen without sound card > (Giovanni Maruzzelli) > 4. Issue with an IVR-menu (Milena) > 5. Re: Aastra SCA Woes, Any Updates? (Brian West) > 6. Re: Aastra SCA Woes, Any Updates? (Chris Chen) > 7. Re: Aastra SCA Woes, Any Updates? (Brian West) > 8. Re: Issue with an IVR-menu (Milena) > 9. Re: Aastra SCA Woes, Any Updates? (Wes Pearce) > > > ---------- Message transféré ---------- > From: k xd <kouxiaodong(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 16:40:47 +0800 > Subject: [Freeswitch-users] How to enable mod_skypopen without sound card > Hi, > > I setup freeswitch in a window server and then startup 2 skype clients, > however when I call registered skype user, I got this message "Contact can > only receive IMs". The server doesn't have sound card. So I guess that when > registered skype users try to transfer the incoming call, because of no > sound card it terminated the call directly. > > Does anyone know how to solve this issue? > > Thanks, > Will > > > ---------- Message transféré ---------- > From: Wes Pearce <wespear(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 10:32:37 -0700 (PDT) > Subject: [Freeswitch-users] Aastra SCA Woes, Any Updates? > > Hello Users, > > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the > office here. I've been trying to get SCA working unsuccessfully. While I > was > digging around, I found this thread: > > http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html > > Which describes my exact problem. SCA works neatly on outbound calls, and > SCA works on incoming calls until the phone is answered... at which point > the call dissapears from the other lines. > > Brian West mentioned this has something to do with Aastras inconsistent > call-info headers. > > I've been speaking with a support representative from Aastra named Brian > Epps, who says their engineers are unaware of any SCA problems. > > Here are the appropriate SIP traces: > > incoming call: http://pastebin.com/QK9K3KL9 > outgoing call: http://pastebin.com/zzxbm9mq > > There are call info headers in both traces. It's just the call-info headers > in the incoming trace look incorrect. > > I'm wondering if there has been any movement on this, or if I've just > screwed something up somewhere. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-t… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Message transféré ---------- > From: Giovanni Maruzzelli <gmaruzz(a)celliax.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 19:52:22 +0200 > Subject: Re: [Freeswitch-users] How to enable mod_skypopen without sound > card > On Thu, Jun 17, 2010 at 10:40 AM, k xd <kouxiaodong(a)gmail.com> wrote: > > Hi, > > I setup freeswitch in a window server and then startup 2 skype clients, > > however when I call registered skype user, I got this message "Contact > can > > only receive IMs". The server doesn't have sound card. So I guess that > when > > registered skype users try to transfer the incoming call, because of no > > sound card it terminated the call directly. > > Does anyone know how to solve this issue? > > A windows server has no way (that I know) to fake a soundcard (as > opposed to the snd-dummy "fake" audio driver in Linux), and the Skype > clients check if there is a soundcard available, and refuse to do > audio if a soundcard is not available (is stupid, but is like that). > > So, your only option is to add a soundcard to the server, a cheap > dongle format USB soundcard would do as well (the soundcard is not > used for anything, but the Skype client check about its existance). > > -giovanni > > > > Thanks, > > Will > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > ---------- Message transféré ---------- > From: Milena <testeador01(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 12:56:46 -0500 > Subject: [Freeswitch-users] Issue with an IVR-menu > Hello, > > Does anybody know why would freeswitch say ... switch_ivr_menu.c:851 Unable > to build xml menu ... mod_dptools.c:1264 Unable to create menu when trying > to call an IVR that was working fine before I updated this morning? ... and > how to fix it? ;) > > http://pastebin.freeswitch.org/13209 > > thank you > > > ---------- Message transféré ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 13:12:43 -0500 > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > The issue is the absence of the extra bits in the call-info header on every > packet involved in the call transaction... but its worse than that if you > want SCA you can only have ONE account on the phone and HOPE and pray it > behaves. I have tried to get Aastra to fix it but as far as I'm concerned > your best bet is to call Polycom and trade all those Aastra's for Polycom's > they have a trade up program and all you have to do is take a baseball bat > and make the Aastra's inoperable or not they seems to have the inoperable > part covered already right out of the box. > > /b > > On Jun 17, 2010, at 12:32 PM, Wes Pearce wrote: > > > > > Hello Users, > > > > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the > > office here. I've been trying to get SCA working unsuccessfully. While I > was > > digging around, I found this thread: > > > > > http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html > > > > Which describes my exact problem. SCA works neatly on outbound calls, and > > SCA works on incoming calls until the phone is answered... at which point > > the call dissapears from the other lines. > > > > Brian West mentioned this has something to do with Aastras inconsistent > > call-info headers. > > > > I've been speaking with a support representative from Aastra named Brian > > Epps, who says their engineers are unaware of any SCA problems. > > > > Here are the appropriate SIP traces: > > > > incoming call: http://pastebin.com/QK9K3KL9 > > outgoing call: http://pastebin.com/zzxbm9mq > > > > There are call info headers in both traces. It's just the call-info > headers > > in the incoming trace look incorrect. > > > > I'm wondering if there has been any movement on this, or if I've just > > screwed something up somewhere. > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-t… > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > ---------- Message transféré ---------- > From: Chris Chen <chris.chen2004(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 14:31:34 -0400 > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > I second that Polycom SCA is working perfectly reliably with multiple > accounts (if you have multiple lines such as Polycom IP 650). > Thanks, > Chris > > On Thu, Jun 17, 2010 at 2:12 PM, Brian West <brian(a)freeswitch.org> wrote: > >> The issue is the absence of the extra bits in the call-info header on >> every packet involved in the call transaction... but its worse than that if >> you want SCA you can only have ONE account on the phone and HOPE and pray it >> behaves. I have tried to get Aastra to fix it but as far as I'm concerned >> your best bet is to call Polycom and trade all those Aastra's for Polycom's >> they have a trade up program and all you have to do is take a baseball bat >> and make the Aastra's inoperable or not they seems to have the inoperable >> part covered already right out of the box. >> >> /b >> >> On Jun 17, 2010, at 12:32 PM, Wes Pearce wrote: >> >> > >> > Hello Users, >> > >> > I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in >> the >> > office here. I've been trying to get SCA working unsuccessfully. While I >> was >> > digging around, I found this thread: >> > >> > >> http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html >> > >> > Which describes my exact problem. SCA works neatly on outbound calls, >> and >> > SCA works on incoming calls until the phone is answered... at which >> point >> > the call dissapears from the other lines. >> > >> > Brian West mentioned this has something to do with Aastras inconsistent >> > call-info headers. >> > >> > I've been speaking with a support representative from Aastra named Brian >> > Epps, who says their engineers are unaware of any SCA problems. >> > >> > Here are the appropriate SIP traces: >> > >> > incoming call: http://pastebin.com/QK9K3KL9 >> > outgoing call: http://pastebin.com/zzxbm9mq >> > >> > There are call info headers in both traces. It's just the call-info >> headers >> > in the incoming trace look incorrect. >> > >> > I'm wondering if there has been any movement on this, or if I've just >> > screwed something up somewhere. >> > -- >> > View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-t… >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Message transféré ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 13:40:14 -0500 > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > The Linksys/CiscoSPA's do too :P And those interop between each other > correctly. > > /b > > On Jun 17, 2010, at 1:31 PM, Chris Chen wrote: > > > I second that Polycom SCA is working perfectly reliably with multiple > accounts (if you have multiple lines such as Polycom IP 650). > > Thanks, > > Chris > > > > On Thu, Jun 17, 2010 at 2:12 PM, Brian West <brian(a)freeswitch.org> > wrote: > > The issue is the absence of the extra bits in the call-info header on > every packet involved in the call transaction... but its worse than that if > you want SCA you can only have ONE account on the phone and HOPE and pray it > behaves. I have tried to get Aastra to fix it but as far as I'm concerned > your best bet is to call Polycom and trade all those Aastra's for Polycom's > they have a trade up program and all you have to do is take a baseball bat > and make the Aastra's inoperable or not they seems to have the inoperable > part covered already right out of the box. > > > > /b > > > > > > ---------- Message transféré ---------- > From: Milena <testeador01(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 14:16:26 -0500 > Subject: Re: [Freeswitch-users] Issue with an IVR-menu > The problem was the empty digits="" that was added when doing tests and the > menu hadn't been reloaded until i restarted freeswitch >_< > thank you anyways :) > > > > ---------- Message transféré ---------- > From: Wes Pearce <wespear(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 17 Jun 2010 12:43:16 -0700 (PDT) > Subject: Re: [Freeswitch-users] Aastra SCA Woes, Any Updates? > > Lame lame lame! > > Thanks though guys, you rock. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-t… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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FS as Media Gateway Only
by Code Ghar 18 Jun '10

18 Jun '10

Is it possible -- and are there any case studies, practical experience, etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP) handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media responsibilities? So when a call comes in, the SDP contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would request FSRTPx for media resources for each new call and add its IP and port in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP only deals with signaling. This way multiple servers could be deployed to handle media responsibilities and only a handful would be required for signaling. In future if there's a greater need for transcoding, etc. all you need to do is deploy a media server and not have to add servers for signaling. This idea came to me because I have come across two proprietary applications that do it this way. They have a SIP component and a media component. You can run both on the same physical machine or you can separate them out into multiple machines. Another way for this could be to integrate FS as a media component to another application's SIP component. A mix-and-match, so to speak. On the flip side, deploy FS as a SIP server and use media capabilities of some other hardware or software application. For example, FS handles signaling and use dedicated hardware for media. A good example of this is illustrated (somewhat) by an image on Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg. Look at the "pooled transcoding". Is FS even designed to be this modular? If so, how can the aforementioned scenario(s) be achieved?

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4 second delay
by Sean Holt 18 Jun '10

18 Jun '10

Hello list, I¹ve been dealing with a particular issue with in-coming calls. Leg A calls into the office, then Leg B (endpoint) picks up call but hears nothing on other side. Wait 4 sec call completes and Leg B can hear the other person. I have Centos 5.4 Latest git build Polycom phones Calling out is not a problem. Not sure how to troubleshoot this issue or maybe there¹s a delay setting in the sip profile that waits for the channel to complete. Thanks for the help Sean

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Playing a voice prompt in a conference
by M. Ranganathan 18 Jun '10

18 Jun '10

Hello, I want to play a prompt ( wav file ) in a conference after two participants join it. What is the best way to accomplish something like that in FreeSWITCH? Are there any examples I can look at? Thank you in advance for any help Ranga -- M. Ranganathan

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Re: [Freeswitch-users] Your Security Best Practices
by Kristian Kielhofner 18 Jun '10

18 Jun '10

What carriers are you using that utilize re-INVITEs? -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces(a)lists.freeswitch.org < freeswitch-users-bounces(a)lists.freeswitch.org> *To*: freeswitch-users(a)lists.freeswitch.org < freeswitch-users(a)lists.freeswitch.org> *Sent*: Tue Jun 15 14:37:02 2010 *Subject*: [Freeswitch-users] Your Security Best Practices Recently I have seen many attempts to break in to VoIP switches I work with. So I thought this might be a good time to discuss everyone's security best practices. If you could relate them to how you implemented them in FreeSWITCH, it would give us beginners good ideas on how to follow the principle of "security first". Let me start off the discussion with some practices we use and have seen in carriers we deal with. Since we only do carrier-to-carrier stuff, it is easy to restrict the IPs where we accept traffic from. This is done using either a dedicated firewall (we use Cisco and pfSense) or host-based (since we use Linux, iptables it is). We restrict ingress traffic on port 5060 (SIP) to trusted sources only. Since a lot of carriers are starting to use re-invites and most do not provide ranges of media IPs they will use, we leave open all UDP ports used by our servers for RTP, usually between 10,000 and 50,000. In this situation, how well does FreeSWITCH cope with attacks? For example, if RTP ports are 10,000 to 20,000, does FS recognize invalid RTP packets and knows how to deal with them? Another scenario is user registration of softphones or ATAs, etc. A practice we use in our lab environment (we will soon be putting servers in production) is to use lengthy passwords, at least 15 characters, mixing alpha-numeric characters. We also do not use 4-digit extensions (default in FS) but use 6-digit. It wasn't started as a security measure but turned out that many attackers we saw usually try 4-digit extensions. So we are safer today (maybe not so in the future when attackers wise up). How do you manage your users, extensions, registrations, etc., from a security aspect? In the same context, we recently saw a flood of invalid user registration requests, at close to 60-80 requests per second, for a sustained period of less than a minute. One proprietary solution we use was unable to handle this flood of requests and dropped 70% of its call before the attacks stopped and it was able to recover. Let's say we use top-of-the-line servers to run FS, maybe two quad-cores with 16GB memory, then should it increase the ability of FS to deal with such unwanted floods? I assume FS is configured with default settings. The real question is: does throwing hardware at the problem solve the issue for FS? And even if it does, is there anything we can do at the FS level to allow it to handle even more flooded traffic on its own, without relying on superior hardware? Then come Man in the Middle attacks. So far I haven't seen a real push in the industry to mitigate these. One common practice is to use prefixes to identify traffic from a source. For example, we can use 999912125550000 where 9999 identifies us to the egress gateway and the rest is the number to call. Of course, since signaling is in plain text someone can analyze this data and then start using the same prefix to send traffic to our egress gateway/carrier. They believe the traffic is from us and charge us for the calls. A second option is to use SIPS but I haven't seen it being used at all. Of course, now we are talking about using TCP instead of UDP, bringing its own advantages and disadvantages. A third option I am starting to see if the use of IPSec tunnels for signaling and RTP is exchanged outside of the tunnel. This works well for carrier-to-carrier traffic but cannot be deployed widely cost-effectively if you do home user to service provider traffic. If you have dealt with any one or all of these scenarios, or have come across others, please share your experiences and solutions with all of us. Of course, if you can also advise how to implement your security best practices in FS it would be just wonderful.

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default SIP registration timeout
by Madovsky 17 Jun '10

17 Jun '10

Is there any xml param for SIP registration timeout ? Thanks F

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Re: [Freeswitch-users] Question about execute_on_answer
by Davide Colombo 17 Jun '10

17 Jun '10

Hi, first of all, thanks for your support! I followed Steven's advice, using a script with exec in answer confirm to play a file and to read dtmf. In this way i'm able to connect caller party with called party, hang-up caller party, move to voicemail caller party, but with multiple calls, phones that don't answer, they continue to ring. To avoid this behaviour i tried in a different way. Calls start with a multiple originate: originate user/1001,user/1002,user/1003 exten-to-ivr When one user answers, he is transfered to an ivr where he can choose the action. In this way caller party remains in ringing state and other phones stop to ring. In exten-to-ivr caller party's uuid is passed to let connection with called party. Davide ----- Messaggio originale ----- Da: "Steven Ayre" <steveayre(a)gmail.com> A: freeswitch-users(a)lists.freeswitch.org Inviato: Martedì, 15 giugno 2010 11:06:37 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Read the "exec in answer confirm" section of http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation I think you would be able to do what you're asking within a script that's executed this way, playing the file and reading DTMF then doing logic based on the pressed key from within the script. Any scripting language should work - so lua etc as well as javascript. -Steve On 15 June 2010 09:46, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi, using group_confirm_file and group_confirm_key i have only one choice, connect called party with calling party (1 dtmf maps with group_confirm_key). What i'd like to do when called party answers, it's: press 1 to connect with calling party, press 2 to hangup calling party, press 3 move calling party to voicemail. Additionally, making parellel dial in bridge, called phones stop to ring only when one of them digits the group_confirm_key and not (my purpose) when one of them answers. These are the reasons to transfer called channel to an ivr context using "execute_on_answer" variable. Best Regards ----- Messaggio originale ----- Da: "Michael Collins" < msc(a)freeswitch.org > A: freeswitch-users(a)lists.freeswitch.org Inviato: Martedì, 15 giugno 2010 0:37:18 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer Aha! You are trying to do some sort of answer verification, correct? You do know that FS already has this feature, correct? Check it out: http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_varia… Make sure that you aren't duplicating functionality before continuing. -MC On Mon, Jun 14, 2010 at 3:12 PM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi, when 1019 answers, i transfer its channel (called channel party) to an IVR context using "execute_on_answer". In this context the called party (1019 in this case) can listen to a message where he can choose to connect/hang-up the calling party using dtmf tones. In the meanwhile i need that calling party remains in a ringing state. Best regards ----- Messaggio originale ----- Da: "Michael Collins" < msc(a)freeswitch.org > A: freeswitch-users(a)lists.freeswitch.org Inviato: Lunedì, 14 giugno 2010 23:46:42 Oggetto: Re: [Freeswitch-users] Question about execute_on_answer I'm not sure I follow what you're trying to accomplish... You call from x1000 to x1019... and what exactly do you want to have happen when 1019 answers? -MC On Mon, Jun 14, 2010 at 8:22 AM, Davide Colombo < dcolombo(a)voismart.it > wrote: Hi all, i'm trying to use "execute_on_answer" channel variable with a transfer before a bridge: <action application="export" data="nolocal:execute_on_answer=transfer myivrpresentation XML default"/> <action application="bridge" data="user/1019@${domain_name}"/> When i call 1019 from 1000 and 1019 answers, i have this situation: 1019 is transfered to myivrpresentation but in console i can see this error 2010-06-14 16:57:51.053528 [DEBUG] switch_core_codec.c:146 sofia/internal/ 1000(a)fs2-devel.voismart.net Restore previous codec PCMU:0. 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2010-06-14 16:57:51.053528 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] while 1000 goes to next action after bridge in the same extension. Is it possible to transfer called channel (when called party answers) to a particular extension and leave caller channel in ringing state? Best Regards _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)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(a)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(a)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(a)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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Aastra SCA Woes, Any Updates?
by Wes Pearce 17 Jun '10

17 Jun '10

Hello Users, I've got a dozen or so Aastra 6757i phones hooked up to FreeSWITCH in the office here. I've been trying to get SCA working unsuccessfully. While I was digging around, I found this thread: http://freeswitch-users.2379917.n2.nabble.com/Aastra-and-SCA-td4995252.html Which describes my exact problem. SCA works neatly on outbound calls, and SCA works on incoming calls until the phone is answered... at which point the call dissapears from the other lines. Brian West mentioned this has something to do with Aastras inconsistent call-info headers. I've been speaking with a support representative from Aastra named Brian Epps, who says their engineers are unaware of any SCA problems. Here are the appropriate SIP traces: incoming call: http://pastebin.com/QK9K3KL9 outgoing call: http://pastebin.com/zzxbm9mq There are call info headers in both traces. It's just the call-info headers in the incoming trace look incorrect. I'm wondering if there has been any movement on this, or if I've just screwed something up somewhere. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Aastra-SCA-Woes-Any-Updates-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Issue with an IVR-menu
by Milena 17 Jun '10

17 Jun '10

Hello, Does anybody know why would freeswitch say ... switch_ivr_menu.c:851 Unable to build xml menu ... mod_dptools.c:1264 Unable to create menu when trying to call an IVR that was working fine before I updated this morning? ... and how to fix it? ;) http://pastebin.freeswitch.org/13209 thank you

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