我正在开发一个项目，其中使用 Sipek Voip 连接到 Freeswitch。情况是这样的：
我有一台Sangoma A400硬盘。我编译了 Windows 版 Freeswitch，现在它可以完美运行。
我还使用 Sipek Voip SDK 创建了一个软电话，它与 Freeswitch 配合良好。
问题是，当我有来电时，我得到的是 mod_sofia，而不是显示呼叫者号码。
我查看了 Sipek，它从 pjsip 获得的只是一个包含 [电子邮件受保护]:5060>。
所以我去了pjsip并试图将实际的电话号码传递给Sipek。我发现有一个名为 pjsua_call_on_incoming 的函数可以处理来电。
它需要一个pjsip_rx_data类型的参数。它有一个字符串字段 (rdata->msg_info.msg_buf)，其中包含整个消息。我尝试替换 [电子邮件受保护]： 5060> 与实际数字，但它没有 影响。
有谁知道如何解决这个问题？
I am working on project in which I use Sipek Voip for connecting to Freeswitch. Here is the situation:
I have a Sangoma A400 hard. I compiled Freeswitch for Windows and now it works perfectly.
I have also created a Softphone using Sipek Voip SDK and it works well with Freeswitch.
The problem is that, when I have an incoming call, instead of showing the callers number, I get mod_sofia.
I looked at Sipek and all it gets from pjsip is a string containing <sip:[email protected]:5060>.
So I went to pjsip and tried to pass the actual phone number to Sipek. I found out there is a function called pjsua_call_on_incoming which handles an incoming call.
It takes an argument of type pjsip_rx_data. It has a string field (rdata->msg_info.msg_buf) which contains the whole message. I tried to replace <sip:[email protected]:5060> with the actual number, but it has no effect.
Does anyone have any idea how to fix this?
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您可以查看此链接来跟踪问题。不幸的是，即使在论坛上，也几乎没有人可以“免费”帮助您解决开源项目。我只是从我个人的经历来说说。我面临着同样的问题，尽管我已经解决了我以前在 SIPEK 中遇到的许多问题，但到目前为止我还无法解决这个问题。
You can check this link for tracking the issue. Unfortunately there are hardly any people who can help you out with Open source projects "for free" even on a forum. I just speak from my personal experience. I am facing the same problem, and cannot figure it out till now, though I have solved many issues that I used to face with SIPEK, all on my own.
我不明白你的问题的根源是在 FreeSWITCH 或 sipek/pjsip 中。
FreeSWITCH wiki 上的此条目可以帮助您调试 FreeSWITCH 中的 sip 堆栈：
http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP
以类似于wireshark捕获的方式。
抱歉，我不知道如何帮助您追踪 pjsip 中 msg_info.msg_buf 的解析/渲染。
I've not understood the root of your problem is in FreeSWITCH or in sipek/pjsip.
This entry on FreeSWITCH wiki could help you debug the sip stack in FreeSWITCH:
http://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP
in a way similar to a wireshark capture.
I'm sorry I don't know how to help you trace down the parsing/rendering of msg_info.msg_buf in pjsip.
在您的拨号字符串中添加
sip_contact_user=xxxx。Add
sip_contact_user=xxxxin your dialstring.